I’m using a Holo Spring 2 Wild edition. I think im treble sensitive and my left ear gets buzzing if the highs are glaring or grainy. Sinc-m and s are alright for me.
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Thanks @jussi_laako for the work on Sinc-L. It does not have that slight exaggeration of room echo but retains all the detail (and more) from Sinc-M. I really liked Closed-Form-M but after listening to Sinc-L for the weekend, its now my strong preference. Could be placebo with 2M taps when doing 16fs scaling PCM->PCM …but it does sound really good.
I would again have to reiterate that my digital setup is optical signal with zero RF/EMI from source components impinging on my DAC (Chord Hugo TT2). So I would like to presume that I am hearing the natural state of the filter …and not the combined CPU/Memory/SSD digital hash that travels meters to affects my DACs transparency.
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Probably silly question but I am not too technical; how do you benefit of the 2M taps with an Optical setup? Is the upsampling not limited to 192khz?
Chord DACs support high speed 705/768khz coaxial interface on dual coax inputs. I use OPTO-DX optical modulation to go electrical-optical-electrical to create a moat around the DAC. So this, in effect, give me the benefits of much higher sampling (4x the 192khz maximum of Toslink) while still retaining the optical gap in an ultra-low-RFI interface. Some users go USB with optical extenders to acheive same …but be wary of the active optical cables (that carry thin copper wires) or the RFI noise receivers devices.
2M taps just refers to the level of ‘polish’ Jussi applies to the audio samples. Approximate analogy: You could use 150 grit sandpaper on a wood surface and call it ‘flat’ …or you could go up to 500 grit for even flatter/smoother.
I use PCM 705/768 with sync-M ou poly-sync-ext2 with LSN5… just for fun, i tried the classical TPDF.
I’m suprised… it sounds more warm. Bass, Organ have more density and sensuality… my imagination ??
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A post was split to a new topic: Desktop 4.6 issues
Updated to HQPlayer embedded on AL (4.18.0) this morning, Sinc-L is a clear improvement (to my ears) over sinc-M for PCM-> PCM for 705/768kHz into Chord DAVE, Sounds smoother, with better separation and clarity
Wanted to add that on my Nuc i7 server, (running at 1.9GHz), even with Sinc-L it is barely breaking a sweat, CPU idling at 98% and temperature at 42%
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Antipodes users - update available for HQP with sinc-l. fYI
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Right, so impressed by HQPlayer I’m now a fully paid up member of the club however I’m kind of still in the dark a bit regarding settings.
@dabassgoesboomboom has already kindly suggested some settings to get going with however with my DAC I don’t know whether I should go the PCM route or the SDM route.
DAC is a Topping D50S so ESS and I’ve successfully got both 768K PCM working (poly-sinc-ext2 and LNS15) working and DSD to 256@12mhz. Current settings are DoP and DSD 48khz are both checked and I’m running poly-sinc-ext2, LNS15, ASDM7 and DSD256 48khz.
Sounds very nice but I’m shooting a bit blind as to whether it’s an optimal set up? Source files are red book CDs ripped to lossless flac or whatever Qobuz chucks at me.
Any pointers would be gratefully received :).
10 points from the jury in Belgium.
Yoiur settings are excellent, but only your ears can tell you what you actually prefer with your DAC: SDM (DSD256) and PCM (768Khz).
I am personally a big fan of DSD, but I am sure I could live with PCM as well (although I might have bought another DAC than my T+A 8 DSD dac). My HQplayer settings are DSD256, poly-sing-xtr-lp, ASDM7.
But what is right for me is far from right for other people.
The only (big) advantage of PCM upsampling is that it needs far less computer power (and so less heat and eventually less noise if fans are involved) than DSD upsampling.
Just listen to your equipment with 1 fix choice for at least 1 month. Than switch to the other setting.
As it will sound different, you will believe in the first couple od days that the newer setting will sound bettter. But, aftert those couple of days, you will really start feeling that you are missing something, of at the contrary you have gained something.
If you really do not hear a ‘qualifying’ difference, stick with PCM.
And again your HQPlayer settings are fine.
Dirk
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Thanks Dirk. I must say I’m deeply impressed by what I’m hearing. The D50S is a hugely competent DAC and for the money pretty much untouchable I think. It was good enough to convince me to sell my Naim ND5 XS2 as performance was very close.
Now with HQPlayer I’d happily put my system up against pretty much any stock streamer/DAC combo regardless of price.
Re: DSD - my pc handles the load fine. Ryzen 5 2600 with GTX1080ti so utilising CUDA offloading. Am going to upgrade to a Ryzen 5 or 7 3xxx very soon though. I have some specialist astrophotography image processing software which is very nearly as resource hungry as HQP so an upgrade will be useful 
And yes, your method of testing; just listening to the system for a month or so makes sense. I’ve often done this when deciding on whether to live with something or not.
The D50s uses asynchronous sample rate conversion internally (the DAC chip).
The Pro-Ject Pre Box S2 (similar price) doesn’t use ASRC.
Short of measuring the D50s analogue output with both PCM and DSD inputs (I haven’t seen any such measurements with the D50s but have for the Pro-Ject) , best to just use your ears to decide PCM vs DSD input
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Good stuff - I’ll have a good listen and see which I prefer. I am liking the current settings very much and am now happy my system (bar a second sub when I find one and a PSX-R2 for my Phono Sig - both of which are minor bits) is now complete.
HQP has certainly provided the last link in the chain…
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Quite useful. Not just for upsampling.
I’m using it with non-Roon inputs and applying digital room EQ (via convolution) to any/all inputs.
Cool…I’ll take a look
Yes room correction is something I need to look into next now I’ve settled on a system
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I wouldnt even know where to start with Convolution
I’m feeling the same way at the moment but trying to do some reading on the matter. All of this is a bit out of my comfort zone it must be said!
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“And that’s all there is to convolution …”
But seriously, convolution is just a way of blending two mathematical functions that is especially convenient if your functions are transforms. Audio input signals can be described as Fourier transforms (combinations of sine waves).
One way of intuitively describing convolution is that it is an expression of how much two functions overlap each other. For example, if the probability distribution of an event F is f(x) and an event G is g(x) then the probability distribution of F AND G (F + G) is the convolution f(x)*g(x).
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In very simple terms, you can use a program like Audiolense (costs money) that will measure the speakers frequency response and impulse response, inside your room at the positions you place the microphone. Typically you will measure at your listening position.
In Audiolense (for example) you can then specific a ‘target’ frequency response (and adjust impulse response if you want) to your liking.
Audiolense then spits out a couple of files that you import into HQplayer that will tell HQPlayer to make adjustments to your music on the fly - real-time adjustments to achieve the target response that you specified, by reducing levels of some frequencies and maybe boosting levels of some frequencies. This is the convolution process. HQPlayer is the ‘convolver’ in this case.
You can also use free software like REW but that can be a little more complex. There is a guide on this forum and lots of Community help if you get stuck, that’s a big positive.
There’s another paid software called Acourate that can be a little complicated compared to Audiolense but perhaps more powerful in terms of features.
Fortunately you don’t need to understand any of the maths involved with the convolution process itself, to benefit from what it can do!
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