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For a long time I just used the poly-sinc filter, then there were some things I liked about the xtr variant. I thought ext sounded good too.

At the moment, in my setup, I have a clear preference for the closed-form. Sounds natural without obvious artefacts and maintains good focus, layering and wide staging. Seems to sound good with all material (to me) and with both my speaker and headphone setups.

All systems/ears are different though.😉


Mac Mini/Uptone Audio JS-2 & MMK > iFi iUSB3 > Benchmark DAC3L > Benchmark HPA4 >  ATC SCM100 ASLT

Headphones: Focal Utopia, Sennheiser HD600; Players: Roon, HQPlayer, Audirvana 3

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Clarification requested, please: When default output mode is set to SDM at 48 x256, and I initiate playback of a DSD64-DSF track, does the oversampling filter apply or just the modulator? I ask because if I've last played an Nx PCM track (Closed-Form-16M), DSD playback won't start unless I first manually switch filter to the 1x setting (ply-sinc-xtr-mp-2s). I really enjoy Closed-Form-16M with Nx PCM but I would think that when tracklist contains a different file type the switch to settings selected in settings dialogue would occur automatically. Guidance appreciated, my settings are below:

Capture.JPG


HQ Player --> iGalvanic3.0 --> Singxer SU-1 --> Matrix X-Sabre Pro --> Feliks Audio Elise Mark II 

Headphone: ZMF Auteur (Walnut) / Headphone Cable: Norne Silvergarde S3

Power Conditioning: Furman Elite-15 PF i

 

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1 hour ago, LoryWiv said:

Clarification requested, please: When default output mode is set to SDM at 48 x256, and I initiate playback of a DSD64-DSF track, does the oversampling filter apply or just the modulator? I ask because if I've last played an Nx PCM track (Closed-Form-16M), DSD playback won't start unless I first manually switch filter to the 1x setting (ply-sinc-xtr-mp-2s). I really enjoy Closed-Form-16M with Nx PCM but I would think that when tracklist contains a different file type the switch to settings selected in settings dialogue would occur automatically. Guidance appreciated, my settings are below:

Capture.JPG

 

Only modulator applies, and some of the settings from "DSD Sources" dialog. So the PCM-to-SDM filter restrictions shouldn't apply. I'll check this case that it is not unnecessarily applying the PCM filter limits.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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3 hours ago, Le Concombre Masqué said:

 

I would upsample all PCM to 192 with Closed Form and shaped dithering ; I'd try none too and play with the dCS filters

 

 

+1 for combination of closed form and shaped dither.

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1 hour ago, Miska said:

Only modulator applies, and some of the settings from "DSD Sources" dialog. So the PCM-to-SDM filter restrictions shouldn't apply. I'll check this case that it is not unnecessarily applying the PCM filter limits.

 

...fixed now for next release...


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Thanks so much to everyone for their help on my DCS Paganini question... One more question.  In HQ ver 4, what are the "Filter Nx" and the "Oversampling Nx" choices for?

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11 hours ago, Le Concombre Masqué said:

The idea is to best the DACs, their modulators, filters, processing power,  etc, by better ones provided by HQP and better run by a powerful computer ; however due to the prestige of dCS, I'd be interested to know what your ears prefer

 

I would let DSD 64 be DSD 64 and check Direct in DSD settings (but if you use convolution filters)

 

I would upsample all PCM to 192 with Closed Form and shaped dithering ; I'd try none too and play with the dCS filters

 

(nowadays) I don't like mp filters

 

 

Hi there.  When you refer to "closed form and shaped dithering" which filter/dither setting do you suggest?  And what, nowadays do you prefer over the mp filters?  thanks much!

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45 minutes ago, evkatz said:

Hi there.  When you refer to "closed form and shaped dithering" which filter/dither setting do you suggest?  And what, nowadays do you prefer over the mp filters?  thanks much!

 

I would suggest you to read ten or so last pages of this thread, there are answers to your questions and lot of other useful information in length..)

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9 hours ago, Miska said:

 

...fixed now for next release...

Thank you!


HQ Player --> iGalvanic3.0 --> Singxer SU-1 --> Matrix X-Sabre Pro --> Feliks Audio Elise Mark II 

Headphone: ZMF Auteur (Walnut) / Headphone Cable: Norne Silvergarde S3

Power Conditioning: Furman Elite-15 PF i

 

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4.0.3 seems to operate better for me, especially in DSD playback from PCM sources.  Previously it would not work sometimes or default back to PCM.  Now everything seems to work fine.

 

One mystery...when i first ran 4.0.3, it locked up on the splash screen.  A reinstall seems to have fixed the problem.  Not sure what happened (MacOS High Sierra).

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14 hours ago, Miska said:

My theory (and ears) tell that filter choice tends to depend on source material.

 

Technically, especially newer source material needs apodizing filters to clean up some of the mess created by half-band digital decimation filters used in ADCs and some production software.

 

Other than that, subjectively I find minimum-phase filters sound good with things like older prog-rock (Pink Floyd etc) recordings and other such multi-track mix studio productions, also with some modern pop tracks. These don't contain any real acoustics at all, only little bit of artificial reverb. (curiously iPhone seems to use minimum phase filters for the headphone output)

 

While I find linear phase subjectively good sounding on recordings made with few microphones in real acoustic spaces.

 

Other than that, recently my preference has been poly-sinc-ext2 which is a linear-phase apodizing filter.

 

Would I be right that these comments apply more or less equally to upsampling to PCM and upsampling to DSD?


Windows 10 PC, Roon, HQPlayer, SOtM sMS-200Ultra, tX-USBultra, Paul Hynes SR4 (x2), Mutec REF10, Mutec MC3+USB, Devialet 1000Pro, KEF Blade.  Plus Pro-Ject Signature 12 TT for playing my 'legacy' vinyl collection.

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thanks for suggesting all the different opinions on filters....


HQ Player (#1) & Audrivana (#2) (wow! love the Apple w/music!!) .. these two software make my system "Amazing!", Purist USB- Benchmark DAC2 HGC (love it!), Purist Audio XLR , ATC SCM25A's (To Die For!) & Focal sub6 . Triode Power Cables with Uber Buss (Yes!) Also enjoy Audeze LCD3 w/"fat pipe cardas."

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1 hour ago, Confused said:

Would I be right that these comments apply more or less equally to upsampling to PCM and upsampling to DSD?

 

Yes, regardless of output format. These reasonings are based on differences in the source content.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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@Miska with the closed-form-megs variants having more taps does that mean they are longer, and thus have more pre/post ringing than the normal variant? 

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43 minutes ago, Yviena said:

@Miska with the closed-form-megs variants having more taps does that mean they are longer, and thus have more pre/post ringing than the normal variant? 

 

Yes, that's what it means. More taps/longer the filter is, longer it "rings".

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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2 hours ago, Miska said:

 

Yes, that's what it means. More taps/longer the filter is, longer it "rings".

 

Ahh okay i was a little unsure about that as I thought longer filters  where independent of number of taps.

But then wouldn't longer filters like xtr/closed mega etc have worse time domain performance or does that somehow improve when upsampling to DSD?

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@Miska Thanks for the explanation  Miska.

 

 I think I remember reading somewhere though in this forum that higher signal speeds like sending/upsampling DSD128/256 improves time domain performance making it not matter so much at these rates, I'm unsure if it was in this thread or not, maybe it was you who wrote that can't remember.

 

On another note I'm wondering is it technically possible to create a filter that is apodizing.but still keeps original samples or is that impossible?

 

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13 minutes ago, Yviena said:

 I think I remember reading somewhere though in this forum that higher signal speeds like sending/upsampling DSD128/256 improves time domain performance making it not matter so much at these rates, I'm unsure if it was in this thread or not, maybe it was you who wrote that can't remember.

 

Most of the limitations come from the lowest rate in the chain. Higher rates allow better construction accuracy though. I'm not completely sure of the context where that has been.

 

15 minutes ago, Yviena said:

On another note I'm wondering is it technically possible to create a filter that is apodizing.but still keeps original samples or is that impossible?

 

That is not possible because by definition the two are mutually exclusive. Intention of apodizing filters is to fix digital filter problems in the source content, and if nothing is changed, no fixing can happen.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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31 minutes ago, Miska said:

 

That is not possible because by definition the two are mutually exclusive. Intention of apodizing filters is to fix digital filter problems in the source content, and if nothing is changed, no fixing can happen.

 

Ah ok, is ADC ringing.really a bad thing though, as I still hear a larger sense of space, details etc with the closed form filter with my adi2/dac1541 so it can't really be so bad, unless it's meant for very badly mastered music.

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5 hours ago, Yviena said:

Ah ok, is ADC ringing.really a bad thing though, as I still hear a larger sense of space, details etc with the closed form filter with my adi2/dac1541 so it can't really be so bad, unless it's meant for very badly mastered music.

 

All ADCs are oversampling and contain digital decimation filter to convert the DSD-like SDM data into lower rate PCM. Many ADCs like the AKM ones in ADI-2 Pro also have selectable digital filters just like DAC side.

 

Or if the material was converted from hires to RedBook for distribution, some software digital filter was likely used instead.

 

How it happens to be depends on what was used to create the source content.

 

Closed-form is interpolator, so it doesn't "filter" anything at all. Totally non-apodizing, you get all the good and bad through. So the results depend a lot on the source material.

 

I personally don't like the aliasing hash modern converters leave at top of the audio band, so I like to use apodizing filters to clean it up.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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On 5/23/2019 at 4:34 AM, Le Concombre Masqué said:

In a word: clarity. 4 brings more micro-details 

I second it... Purchased the license straight away.. Can never go back to version 3 again.. 

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2 minutes ago, PPk said:

I second it... Purchased the license straight away.. Can never go back to version 3 again.. 

I've been waiting to purchase version 4 until all the bugs are figured out, specifically with Server 2016. Is anyone having problems using version 4 with Server 2016 and Roon?


12TB NAS >> i7-6700 Server/Control PC >> i3-5015u NAA >> Singxer SU-1 DDC (modded) >> Holo Spring L3 DAC >> Accustic Arts Power 1 int amp >> Sonus Faber Guaneri Evolution speakers + REL T/5i sub (x2)

 

Other components:

UpTone Audio LPS1.2/IsoRegen, Fiber Switch and FMC, Windows Server 2016 OS, Audiophile Optimizer 3.0, Fidelizer Pro 6, HQ Player, Roonserver, PS Audio P3 AC regenerator, HDPlex 400W ATX & 200W Linear PSU, Light Harmonic Lightspeed Split USB cable, Synergistic Research Tungsten AC power cords, Tara Labs The One speaker cables, Tara Labs The Two Extended with HFX Station IC, Oyaide R1 outlets, Stillpoints Ultra Mini footers, Hi-Fi Tuning fuses, Vicoustic/RealTraps/GIK room treatments

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Just tried 4.0.3 (on latest Mac OS .. 10.14.15) but have found that I can no longer adjust the volume of the DAC from HQPlayer (or from Room).  Everything else seems to work ok.  Anyone else seeing the same issue?  Am using NAA (microrendu) over Ethernet as the back end.

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