[bogi]

Nearly every other filter -- apodizing or not -- does not share this trait. With those other filters, when you input a sample value x at audio stream position t, and you inspect that sample value t at the output (so awaiting the latency of the filter), the value is almost certainly not x anymore

AFAIK all filters which are not marked as apodizing in the HQPlayer filter table preserve original sample values. Only filters, which are marked as apodizing may change original sample values. Mostly it does not happen. @jlaako may confirm this ...

Preserving original sample values is not specific only for closed form filters. Schiit mentions these because they implemented them in their product.

 

[Ghoostknight]

I think we are talking about two different things. I understand that apodizing filters can alter samples (for the better) and know that the manual clearly marks which filters are apodizing. Different from apodization, I was referring to this blurb from Schiit:

"The math involved in developing the filter and calculating has a closed form solution. It is not an approximation, as all other filters I have studied (most, if not all of them). Therefore, all of the original samples are output. This could be referred to fairly as bit perfect; what comes in goes out." (emphasis mine) This and more under the FAQ at https://www.schiit.com/products/yggdrasil.

Nearly every other filter -- apodizing or not -- does not share this trait. With those other filters, when you input a sample value x at audio stream position t, and you inspect that sample value t at the output (so awaiting the latency of the filter), the value is almost certainly not x anymore. What's better is another matter, but this in=out thing and only interpolating in-between is something particular to that closed form filter.

hmmm maybe thats why bass seems most boosted compared to any other filter (im talking sub bass here)

most tweaks have a particular change thats most heared in bass and high frequencys (atleast those two are always determining factors for me on my two way speakers, it might be different with 3-way)
and usually positive impacts in bass usually sound "larger" with more impact... thats definitely the case with closed-form

i might revisit closed-form today again and maybe turn my bass eq a bit back to a audible similar level to sincMGa... it might be a good reveal

closed-form sounded great, just the huge bass impact i never heared before of this kind made me kinda question it, beside the bass it was quiete comparable to the other 1M tap filters

 

[Louisiana]

It seems like this could be my new filter for PCM:

 

[roderickvd]

AFAIK all filters which are not marked as apodizing in the HQPlayer filter table preserve original sample values. Only filters, which are marked as apodizing may change original sample values. Mostly it does not happen. @jlaako may confirm this ...

Again, I’m not talking about apodization.
Any other IIR and FIR filter (not only in name, so including the (poly-)sinc family) do not have output samples equal to exactly their input samples with only interpolation in between.

Edit: Actually, I take this back. Given you’re right that sinc filters preserve the input samples exactly, I seem to have fallen into Schiit’s marketing claim.

 

[BShaw]

Hi everyone. I installed HQP two days ago, and I'm having issues with delayed command response and dropouts, and wonder if anyone has encountered this, and how it was resolved. Here's my setup:
2018 mac mini, i3 8gb RAM>Gustard N18 switch/fiber converter (fiber out)>Teradak Fiber converter (fiber in)>Gustard R26 NOS off>Khozmo pre>Sublime K231 active xover.
My HQP settings are - 768k sample rate, poly-sinc-gauss-xla, LNS15.
Inputs are set at (none) for backend, default buffer time and DoP for the SDM.
Outputs are set at Network Audio Adapter for backend, SDM pack (none), and PCM default.
My best guess is I need more RAM on the mini, so I've ordered 32 gigs just to be sure.

Have I made a mistake in the settings? Any ideas? I'll continue to read the thread when time permits, but just wanted to ask directly. Thank you!

By the way, I only tried a few filters, but the poly sinc Gauss XLA is amazing in my system - it's like inserting all the best aspects of tubes (holography, spaciousness, depth of stage, realism). In my biamped system, my horns are playing from 600 hz up and tube-amped, so this is where I'm hearing all this goodness.

 

[jlaako]

I agree that a basic description how to use this scenario is missing in the manual. @jlaako

In near future, I will ask Aapo (the person doing key updates and such for Embedded) to make some guides to the web pages about how to set such things up. If I write such, I will always accidentally omit some important parts because I'm just too familiar with the dirty details...

you need to paste "audio:default/0/2" to input URI line and start playback

One important detail here is to press enter after having such input (this URI is now also listed by default on the drop-list), to "load" it. If you already had that input and your playlist looks empty, clear the entry and re-select that URI and then press enter. Once it appears on the playlist / playback queue, it is ready to start playback from. This URI entry is trying to be conceptually similar to the URL entry of a web browser.

 

[Ghoostknight]

i might revisit closed-form today again and maybe turn my bass eq a bit back to a audible similar level to sincMGa... it might be a good reveal

First test song was with many apodizing errors, no thanks, i stay with sincMGa

apodizing errors that dont get corrected just dont sound right imo it kinda sounds crackly and edgy

 

[bogi]

i everyone. I installed HQP two days ago, and I'm having issues with delayed command response and dropouts

Delayed responses are how HQPlayer works. Short filetrs bring short delays, because they need to read sugnificantly less previous samples to compute sample at a given time. Some delay is added by NAA output buffer and a small amount by audio driver of NAA conatined in Gustard R26.

Dropouts may come from many reasons. If the dropouts are regular and not random, the usual reason is insufficient CPU frequency for the filter/modulator combination tried. There may be other reasons too, like thermal throttling, USB cable issues etc., but you didn't provide enough information about your issue.

 

[Currawong]

It is not specific to whether you listen with loudspeakers or headphones. But more on signal properties of the source content, and how it was created. Minimum-phase filters are more natural, since they don't have pre-ringing / "pre-echo". But this is emphasized only on content with regular transients with high frequency content. Like rock for example.

There's a great misunderstanding about ringing in digital filters. What people see as "ringing" when an impulse response is run through a filter is the result of the effect of a non-audio signal -- basically a single, non-zero sample amongst silence. A digital filter of sufficient length will not "ring" when fed with a bandwidth-limited (in-band) signal.

On the other hand, a leaky filter will cause aliasing in the in-band audio (MQA's filters, and similar short, minimum-phase filters are good examples).

Troll Audio's guide to filters and ringing.
Rob Watts on filtering and ringing

apodizing errors that dont get corrected just dont sound right imo it kinda sounds crackly and edgy

No, that'd likely be inter-sample overs. HQPlayer's output should be set at -3dB to ensure this doesn't happen.

 

[Ghoostknight]

No, that'd likely be inter-sample overs. HQPlayer's output should be set at -3dB to ensure this doesn't happen.

it was

 

[plumpudding2]

There's a great misunderstanding about ringing in digital filters. What people see as "ringing" when an impulse response is run through a filter is the result of the effect of a non-audio signal -- basically a single, non-zero sample amongst silence. A digital filter of sufficient length will not "ring" when fed with a bandwidth-limited (in-band) signal.

On the other hand, a leaky filter will cause aliasing in the in-band audio (MQA's filters, and similar short, minimum-phase filters are good examples).

Troll Audio's guide to filters and ringing.
Rob Watts on filtering and ringing



No, that'd likely be inter-sample overs. HQPlayer's output should be set at -3dB to ensure this doesn't happen.

Are you trying to add context or are you trying to educate Jussi on digital filtering?

 

[gammi]

Can anyone explain the differences between poly-sinc-xla and Sinc-MGa? I see MGa is a variant of xla. I also notice the description for xla references SDM whereas the MGa description references PCM. Is MGa better suited for PCM and xla better for SDM/DSD?

I also find it interesting that xla is "xtra long" but still listed as benefitting transients. How is it seemingly giving the benefits of both a short and long filter at the same time?
I've attached the manual description for both as reference.

 

[plumpudding2]

Can anyone explain the differences between poly-sinc-xla and Sinc-MGa? I see MGa is a variant of xla. I also notice the description for xla references SDM whereas the MGa description references PCM. Is MGa better suited for PCM and xla better for SDM/DSD?

I also find it interesting that xla is "xtra long" but still listed as benefitting transients. How is it seemingly giving the benefits of both a short and long filter at the same time?
I've attached the manual description for both as reference.

All Gaussian filters have optimal time domain performance given the filter length. So a long Gaussian filter isn't necessary better at transients than a different shorter filter, it's just the best you can possibly do at that length.

 

[bogi]

I also notice the description for xla references SDM whereas the MGa description references PCM. Is MGa better suited for PCM and xla better for SDM/DSD?

AFAIK no such thing exists. That looks rather like a result of copy&paste of filter descriptions between PCM and SDM tables.

I also find it interesting that xla is "xtra long"

xtra long within that gauss filter family.
You can get an idea about filter length by noticing delay when you start playback. With gauss-xla it starts almost immediately. Switch to sinc-Mx and try the same... While upsampling (interpolating), HQPlayer needs to read number of samples equal to filter length to compute a new sample value.

 

[BShaw]

AFAIK no such thing exists. That looks rather like a result of copy&paste of filter descriptions between PCM and SDM tables.


xtra long within that gauss filter family.
You can get an idea about filter length by noticing delay when you start playback. With gauss-xla it starts almost immediately. Switch to sinc-Mx and try the same... While upsampling (interpolating), HQPlayer needs to read number of samples equal to filter length to compute a new sample value.

I'm using gauss-xla and there's a 5-6 second delay across the board. At 768k there are constant dropouts (3-4 per minute), at 384 almost none, but the lag is still there. I'm hoping/guessing it's a RAM issue because my mini only has 8gb and I have 32gb on order. The mini is a 2018 with a 3.6Gb quad core i3, I would think this is plenty of cpu power for PCM? There are no USB connections in my chain, fiber from the network switch, and a very high quality silver LAN cable directly into my DAC from the fiber converter, also very high quality with LPS. Maybe this provides the missing info you referred to above in post 938 Bogi?