I’ve been trying poly-sinc-short-mp, and it’s great.
What technically is supposed to be the difference between this one, and the xtr-short-mp?
Thanks.
Can I recommend the Lampizator DACs with pre-amps/volume controls? I have the Baltic3 (there’s now a v4) and it works really well with HQP. The sound is truly amazing through a Boulder power amp and Focal Mini Utopias.
Just two different types of filters with slightly different design parameters as well.
I tend to agree, gauss-xla sounds quite spectacular with all genres with the 7EC-super modulator. Gauss-long might suit some songs better but if I could only have one I’d without a doubt want the -xla, because it never sounds like I’m missing something/need to fix something. Sometimes I switch to other filters just to remind myself what a good overall package this filter is.
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After running sinc-MGa @ 512 with 7EC-light 512+fs for a few months I’ve been converted back to sinc-L PCM. MGa with DSD sounded detailed, layered and spacious but what changed to me was demoing a set of high end Focal speakers. I’ve been chasing the transparency and especially the sense of impact of that sound ever since.
A sense of veiledness and bluntedness crept up on me and PCM sinc-L is currently the antidote, incredibly transparent and punchy sound.
It is super interesting how our ears and preferences are constantly in flux, I’m sure in a few weeks my current combo will be harsh and 2D sounding and i’ll go back to DSD. Thankfully with HQPlayer i’m not buying a new dac every time this happens but I can simply tune the sound to my liking! 
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I get what you’re saying, I was surprised that you were so happy to switch to PCM but then I remembered that (correct me if I’m wrong) you’re using a Holo Audio May DAC. Still, this actually makes me perplexed as I’d expect a DAC like that one (which to me appears to be optimised for both PCM and DSD) to be equally good with both formats. But it might be due to the modulator that you’re using. Try ASDM7EC-super. When I want punchy/hard-hitting sound I usually go for Sinc-Mx (sometimes Sinc-L) but still stay in DSD (I have a Delta-Sigma DAC, though).
Hehe, it makes me question my own preferences sometimes. As much as I often go for the gauss filters, tonight is my Sinc-Mx night as I was longing for the tangible, full-bodied (even if sometimes a little sharp) sound.
PS My tonight’s romance with Sinc-Mx didn’t last too long as I found edges of sounds (transients?) to be artificially sharp in the long term and my ears got tired very quickly. The initial impression of that filter are almost always good (it’s sharper so to my ears can come across as being a higher resolution which can result in a “wow” moment) but the experience just never lasts too long. I think my ears are busy dealing with each sharp hit and are thus distracted from the music as a whole.
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I indeed have the Holo May, wonderful dac in both PCM and DSD! I know that Jussi has shown that DSD output is cleaner than PCM on the Holo Spring 3 which he owns, and it is a big shame he’s never had the chance to measure the May because I have a feeling PCM performance might be significantly better.
The PCM 44.1 measurements from Stereophile show how big the difference between Spring and May really is:
To my ears going from DSD to PCM on the May you trade space and a bit of layering for impact and immediacy. Equally good to my ears, just different. I think whether the long sinc filters are either “too edgy” or “just right” also depends on amplifier + headphones/speakers. My current favorites Susvara and DCA Stealth have a reputation for being a bit softer sounding which I can imagine that compensates a little.
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Just to confirm, if I want to completely bypass all hqplayer processing (to do a little A/B comparison):
filter = none
shaper = none
volume= 0dB
Is there anything else that needs to be set?
Thanks!
I don’t know what they did with Spring3, but here’s what my Spring 3 says on the same measurement:
While the same at DSD256 from HQPlayer with ASDM7ECv2:
Main issue with PCM is that the sampling rates are insufficient, while increasing rates make settling times suffer which in turns makes performance suffer. So you have lose-lose situation. What ever you do, performance is limited the by architecture.
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@jussi_laako Does controlling volume via HQPlayer affect dynamic range in any way? I usually keep it as low as -20dBFS because of my class A amplifier being run in bypass mode (bypasses internal volume control chip - big difference). I suspect that any impact on the signal will be minimal (especially compared to doing this in the DAC, which I partially do also, or in the amp) but I’m wondering if there’s a threshold which should not be exceeded for achieving best results in terms of fidelity to the original signal.
I wonder if that Stereophile measurement even is for Spring 3? Could it be Spring 2?
Here’s Spring 3 measuring pretty much perfectly for an R2R DAC.
EDIT:
I quickly googled it and it seems that those Stereophile Spring measurements were for the first generation Spring, so not even the Spring 2.
ah the article said HoloAudio Spring DAC “Kitsuné Tuned Edition” Level 3 which is of course not the same as the Spring 3 Level 3… my mistake! Thanks for the correction.
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Dynamic range decreases exactly by the volume setting with digital volume control, dac noise remains the same while signal level drops so signal to noise ratio (which is dynamic range) has to decrease too.
However if your dac has 120 db dynamic range and your amp has 100 db dynamic range, you can safely use -20db digital volume control as in either scenario you’re limited by the dynamic range of your amp. Even if you use more, it is important to note that analog volume control has noise too, John Siau of Benchmark made an interesting comparison between using digital volume control and their LA4 preamp and it gave a S/N ratio improvement of only 10 db:
In short it depends on the rest of your gear 
Thanks for the reply, as another poster pointed out I confused Spring 1 level 3 and Spring 3 so that explains the discrepancy.
I’m interested in these architecture limitations of PCM. I assume that by insufficient sampling rate is that images at 1.5 Mhz are not completely removed by the analog filter and still cause a bit of intermodulation distortion in the audible band?
I’ve never heard oft the settling time limitation. Can you explain why DSD doesn’t suffer from that or is that something you need to take an electrical engineering course for?
It depends on how the digital volume control is implemented. And I dare to say that it’s state of the art with HQP.
Here’s a good white paper about Roon’s integrated volume control:
Thanks, but according to what @patouskii posted above, if HQPlayer’s volume control is as good as or better than Roon’s then decreasing volume by -20dB would still leave me with a higher dynamic range than my DAC is capable of reproducing. So it shouldn’t have any impact at all.
My strategy has been to leave my amp in medium gain (at low gain it attenuates the incoming signal more, thus decreasing its dynamic range, as I believe it to be the case) and then volume-control via HQPlayer and my DAC - with the emphasis on the former, however I need a hardware safety net to protect my ears if something goes wrong, hence I would not consider using software alone. If anything, I’d maybe look at a better amp.
I’m wondering how much volume attenuation can be applied in HQPlayer before we even start approaching the area where a 16-bit song’s dynamic range might suffer. I’d think that it’s likely much higher than the -20dB I’ve been using.
Good DSP indeed leaves digital dynamic range intact by lowering the digital noise floor to figures like -180 dB. However in my above post I was not talking about digital noise floor but analog, which is signal independent. Even state of the art dacs have an analog noise floor of -120 to -130 dB. That means a 0 dB signal will have full dynamic range, and a -20 dB signal will have 100-110 dB dynamic range.
Again this is only a problem if your amp has more than 100-110 dB dynamic range.
Gotcha, thanks. My amp’s max dynamic range is around 20dB lower than my DAC’s so in my case even using some attenuation in the DAC will effectively make no difference as the amp will be the bottleneck. Although I’d bet that a lot of the music doesn’t even approach the DR levels where this would be a genuine worry.
Funny thing I’ve noticed is that a lot of even expensive amps effectively operate at around 65-70dB dynamic range once comfortable listening volume has been applied (I have mine in bypass mode so I’d like to think that it’s operating at around 95-100dB). Which makes me think that either most music doesn’t exceed those numbers or that the flattening of DR is subjectively pleasant to some listeners (easier to listen to?).
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I have a question about the modulators in Hqplayer.
I find when I use poly-sinc-long-mp for 1x(44.1/48) I like the sound with ASDM7EC-super 512+fs.
But when I’m playing Nx hires using poly-sinc-gauss-hires-mp, I prefer it with the sound of ASDM7ECV3.
Would it be hypothetically possible to add that option to Hqplayer for a future release…to have a separate modulator choices for 1x and Nx in setting?
Thanks.