copy_of_a

out of interest: how much attenuation do you have to apply in HQPs volume control to reach moderate listening levels like 60-70 dBA?

yet corrections to the bass spectrum with linear phase filters are the most problematic. The lower the frequency, the longer the filters impluse response. For the low end linear phase filters should be really broad (low Q) to avoid strong ringing effects.

cool! Many thanks for sharing the graph!! So it's -200dB attentuation for ps-gauss yes, perfect choice (if the source is technically flawless - which is typically the case with classical records)!

Agreed! Regarding Timbre ps-xtr-short-lp stands out! For me personally it is the strongest amongst the HQP filters timbre-wise. Since ps-gauss has a bit more "air" (at least with transient rich music) it sounds a bit "lighter" while ps-xtr-short-lp sounds somewhat denser / meatier. IMHO that's a perception of the balance in the whole spectrum - bass, mids and treble are essentially the same with both filters, only the top end is different (see attachment*). I can very well well understand that you lean towards ps-xtr-short-lp! :-) to me as well ________________________ * the screenshot shows the audio-content of the difference between ps-gauss and ps-xtr-short-lp. Source was a snippet from a loud prog rock song upsampled from 44.1kHz to 88.2kHz with both the respective filters.

My 2 favorite filters :-) both have similar length and roll off (ps-gauss track above / white // ps-xtr-short-lp track below / blue). AFAIK also attentuation is similarly high (-300dB ?). Although ps-gauss rolls off a tad earlier it sounds more open / airy - due to the algorithm transients are better preserved. ps-gauss is a great allrounder (incl. classical music, although ps-gauss-long and especially ps-gauss-xla could be preferred). ps-short-xtr-lp has a slightly more intimate sound (with headphones you literally get sucked into the music). Both sound very accurate and super detailed. Personally I use ps-gauss as my preference with speakers. ps-xtr-short-lp is my preference with headphones (except for classical music). IMHO

to compensate wall bounce (half and quarter space placement compensation) you need a 12dB/oct slope without resonance. So either S=1 or Q=0.709

to enter in the matrix pipeline fields: iir:type=lshelf;f=200;s=1;g=-6 Or to copy into a text file: Filter: ON LS Fc 21 Hz Gain -6.00 dB S 1 edit: Miska was ahead of me :-)

Agreed. However, if you still need (or want) to lower the deepest bass (for instance to gain more precision with smaller speakers or so) you can use a Low Shelf instead of a Low Cut (High Pass). Shelf filters in comparision exhibit way less phase shifts and therefore do not push that much energy into the frequencies above the corner frequency of the filter. See attachments: red = HPF (12dB/oct at 20Hz), green = Low Shelf -6dB (12dB/octb at 20Hz). 1st = frequency; 2nd = group delay

That‘s exactly what I did! Whatever…

There are 2 new eartips from Azla that work perfectly in conjunction with the Kato (and similar IEMs). A new Sednafit Xelastec version that is tighter on the IEM's nozzle, provides even better isolation and comfort and is far less prone to collect dust and dirt like the original Xelacstec version. A great update on an already very good eartip. https://store.azla.co.kr/products/sednaearfit-xelastec-2 And the new Sednafit Origin that slips somewhat deeper into the ear canal. Isolation is top notch and therefore it reproduces more bass. https://store.azla.co.kr/collections/sednaearfit-series/products/sednaearfit-origin-1 I love the "Origin" on the Kato while I use the Xelastec 2 on the Letshuoer S12. Well worth to try out ...

Here's a test track from the Sony site - https://helpguide.sony.net/high-res/sample1/v1/en/index.html I hope it's free to use for this purpose. The file contains different sections: I've applied 12bit, 16bit, 20bit and 24bit TPDF dither to it, downsampled from 96kHz to 44.1kHz and applied additional 24bit TPDF dither. Can you spot the timecodes where I've made the cuts from one bit depth to another? How relevant is the difference between 16bit, 20bit and 24bit for playback? dither_comp.flac

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There’s no dictate. There are facts and there’s knowledge provided by experts. We poor users can or can not appreciate it and act accordingly or not. You can do with your DAC and feed it with whatever you want. What you can’t do is to create alternative facts based on your subjectivity (which is based on a lack of knowledge). We have enough of this BS on planet earth these days. ___________ I wonder why 90dB (15bit) effectively usable dynamic range is of any concern for you. Do you have an iPhone or similar? Get an app to measure the sound level through your phones microphone. Measure the sound level of „silence“ in your listening room (the noise floor of your listening room so to say). Now imagine how loud you have to play music before the noise floor of your DAC would actually be audible. And even with headphones 90dB are at least ‚ok‘.

if you look at the entire measurement set it barely reaches 16bit. https://www.l7audiolab.com/f/gustard-r26/ If I had one I‘d probably set HQP to 15bits.

Well, as I said above: "Bits should match the actual capabilities of the DAC in PCM mode (unless the audio path is not direct and the DAC further procsses/modulates the signal). " But you use HQPlayer for upsampling to bypass the DACs internal processing (as far as it is possible). In this case you should take care that the selected Bits match the actual capabilities of the DAC.