Which HQP Filter are you using?

Hey! Thanks for the reply!

Sorry for the delayed acknowledgement as I had been busy trying to find a place to move and settling in.

Finally, relaxing with new convolution files I was sent and wanted to find out if someone could direct me on the difference between using 192 vs 384 mono wav files. My X26Pro I believe is set to 32 in the settings as well as Roon.

Also, if adding a bass shelf of 4db the already add convolution matrix, is it ok to still add the shelf to the matrix pipeline with a -4db gain adjustment?

You can combine convolution and parametric EQ, like adding a shelf if needed. Items are just comma separated on the pipeline spec.

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From my understanding you should use the highest possible available. I went with the 384k (even if the highest PCM source you are playing is 192k there is no harm in this)

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@jussi_laako :
I was reading this:
And I was wondering how to choose the best setting on HQPlayer for my Pegasus (Pontus like for signal processing).
I’m going through an iFi Zenstream, so limited to DSD256x48.
I’m still a fan of poly-sinc-gauss_long or hires but for the shaper,? “…Denafrips products use FPGA to convert 1-bit DSD to 6-bit (7-level) DSD…
Still ASDM7ECv2… or AMSDM7 512fs which seems more " fluid ".
Given the architecture described by Denafrips, the “NOS” setting does not come into play for DSD (?)

NOS setting on the Pontus II is only used for incoming PCM.
The manual says:

"3.5 NOS/OS
The PONTUS allow the user to change the sampling mode on the fly.
NOS, as the name suggested, does not over-sampling to digital input data.
In OS mode, the PCM 44.1kHz or 48kHz based audio data are up-sampled to the maximum rate of PCM1411.2 or PCM1536. There is no up-sampling of DSD audio signal.

The PONTUS is equipped with 24Bit R-2R DAC to decode PCM data stream and 32 steps FIR analogue filters hardware decoder to decode DSD data stream. These designs guaranteed the PCM format can be perfectly decoded, at the same time, the DSD format can be perfectly decoded as well. It is rare in the currently market that a R-2R DAC can hardware
decode both the PCM and DSD formats."

I don’t fully understand the 32 step DSD part, anyone?

Its a 32 step FIR (Finite Impulse Response) filter, which in theory is how @jussi_laako DSC-1 works for DSD conversion as well. (Direct 1-Bit DSD, 32 tap FIR converter) but Denafrips R2R DACs are a bit of a black box so who knows what the secret sauce is

I like the content on a low-bitrate stream PlanetPootwaddle (128k MP3) and continue to be impressed with poly-sinc-mqa/mp3! It makes it more than listenable :wink:

With other filters the stream can sound quite gritty and crunchy in the upper frequencies. Would I like a higher bitrate, of course, but this is all they stream…

Thank you Jussi for all your talent, work and persistence improving my listening experience so much.


Could I get some recommendations on what filters to use for a DAC that is using TI PCM1794A chip?

I have a DAC with this chip made by a local hi-fi components maker, so very few people have a DAC like that and probably none of them use HQPlayer :slight_smile:

It can do 24/192 PCM maximum and I am feeding it with RPi4/Allo Digione Signature acting as NAA endpoint to the DAC’s AES/EBU input. I haven’t tried all the filter combinations yet, but I keep using and coming back to 1x/Nx poly-sinc-ext2 and NS4 combo.

Remember to set DAC Bits to 24. You can use the table in manual for some filter suggestions. But your current filter choice is also good. Another one you can try is 1x=poly-sinc-gauss-long Nx=poly-sinc-gauss-hires-lp and TPDF and Gauss1 dithers.

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Thank you Jussi. Yes, I have set the bits to 24. Will need more time for listening, but after a quick listen your suggested filters probably sound a bit better than poly-sinc-ext2. Sounds less stressed in some situations and a bit more balanced, relaxed and natural overall. I think this may become my new favorite combination! Thank you!

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@jussi_laako found a strange bug … 88/176 → DSD256x48 (non-integral ratio) works fine with poly-sinc-gauss-hires-lp (linear phase) but with poly-sinc-gauss-hires-ip or -mp it takes roughly 40 seconds of spinning beach ball before it starts playing! This is on a Mac Mini M1 on 4.19.0. Integral upsamples (48, 96, 192) work fine. And all source rates → DSD256x44.1 work fine, integral ratio or not. It’s only with non-integral ratio to DSD256x48 with the -ip or -mp versions.

I’m sure Jussi will look at if it’s a bug, but a quick question, why not enable ‘adaptive output rate’?

This way you will get integer ratios for all rates.

Because I get pops when changing between DSD256x44.1 and DSD256x48 rates. And then i decided that if my DAC can do both then why not just stick with one. So I went with DSD256x48 and ran into this weird issue. It’s not the end of the world but seemed odd.

What about if you fix output to DSD256x44.1 and try 96/192k music.

Same issue?

It’s not a bug, in that case the initialization just takes longer because of the extra work.

Interesting. I am curious why it only happens with -ip and -mp and not -lp. Also curious why it doesn’t happen with non-integral 96/192 → DSD256x44.1… it’s such a drastic difference with that particular permutation compared to all others.

There’s extra computations needed for other forms of filter than linear phase.

Ok thanks. Why doesn’t it happen with non-integral 96/192 → DSD256x44.1?

That case is simpler. The time it takes to initialize is directly related to complexity of the case.

Initializing from 48k to 44.1x256 also takes 4x more time than doing the same from 192k to 44.1x256.

Hi, which filter for PCM would you sign as most spatial, airy or with most separated instruments? Thanks
My settings are this: