phofman

Sorry, now I see the highlighted SPDIF input transformer. Well, as it was already said here, the purpose of SPDIF transformer is not to improve the waveform but to provide galvanic isolation of two physically distinct devices possibly powered from different sources. Every transformer being an LC circle inevitably damages digital signal waveform - the originally straight edges are always "smoothed", that is no rocket science. Another point in the answer - no PLL used in the clock. Of course there is no PLL in the clock generator, nobody talked about PLL. PLL was not part of the original crystal on the non-modified card either. There is a PLL in the USB chip itself but not much can be done with that. I still have not read a single technical argument from the author about what positive effects the transformer between the clock module and the USB chip both powered by the same supply provides. And that is certainly not about anyone's english.

Hey angry man - using your own words - "Just to follow up... When did I say that?" I never make sound judgements as they indeed do require the DBT which you will never dare to take. Perhaps you should calm down and read properly first.

If you are happy with such an answer then good for you. To me it sounds like a random compilation of technical terms. And I do not see any relevance to the DAC linked, its transformers are used for I/V conversion behind the DAC stage. The clock waveform is exactly what will be ruined by the slow transformer inserted into path of the 24MHz clock signal.

If I see correctly out of us three you are the only one commenting on the sound.

There is no galvanic isolation, the two boards share the same power supply. Very different from the SPDIF link situation where two physically distinct devices are hooked with, likely to suffer from a ground loop without the transformer.

It is not my money which buy this , I am not to approve or deny. Just every engineer tries hard to keep the clock signal as precise as possible: minimizing routes, placing the clock generator right next to the chip, shielding, keeping the impedance of the clock source low etc. A bit like the original card - the crystal is right next to the chip. What this "mod" does is the complete opposite - sharp edges of the clock board are being blurred by a bandwidth-limiting transformer, adding further jitter to the signal by catching all the PC EMI/RFI noise induced in the huge transformer (compare its size to SPDIF signal transformers often found on PCI soundcards), routes from the clock to the chip many times longer than the original connection, plain unshielded long wire carrying the signal. But I understand - a product for audiophiles must boast an audiophile-grade transformer, spells like that sell. Interestingly, the professional SOTM card delivers without any transformer. And no wonder.

I have already posted it here, nevertheless: This Browse for Products | CPC (i.e. a PCI-e 1.1 x1 card) + this HIFI with a crystal of different frequency + a transformer ruining waveform (i.e. quality) of the output clock signal = SHOPPING AREA: AUDIO GRADE USB 3.0 PCIE CARD Look at the USB card specifications for your card details. I could not find any user manual online for the model PSG90847, probably some OEM for the Pro Signal brand.

PRO SIGNAL - Interface Card 2 Port Pci-E Usb3 0 - PSG90847 HIFI plus bandwidth-limiting transformer catching the PC EMI/RFI noise to add uncertainty and jitter to the originally sharp edges of the clock signal generated by the the added clock module.

Well in such case I am afraid I cannot help you.

I would be surprised if your game did not use an audio chain resamples to native soundcard rate automatically. If it does not and talks to the soundcard directly (which I very much doubt), you are out of luck with this card. Did you try unchecking the exclusive mode controls on your screenshot?

Your chain will resample automatically to/from 44100, the only native rate of your soundcard. Why do you need to see that option switched to 48kHz?

The device offers only stereo 44.1/16: http://alinux.tv/fender/MustangII.txt

Excellent, very good to know. Perhaps your board does not suffer from the RaspberryPi woes. Mixer in this case is the part of the chain performing volume control. You can either use the volume control facility provided by your HW (the driver will operate the "sliders" of your soundcard) or by MPD itself (software). Vast majority of soundcards uses digitital volume control of its codec chip therefore the final result - samples entering DA conversion - is identical to software volume control.

Thanks for your insight. I looked up the problems - those 8 thousand IRQs per second, no HW support for proper interrupts, rotten unsupportable drivers without HW documentation... it is really a dead-end street. Thanks for the wandboard pointer.

Quality resampling can be very CPU demanding. I am not surprised your arm core cannot cope with it without xruns, even modern desktop CPUs can choke on this task. Playing native 192kHz puts significantly less load than resampling.