DoP Mac M1

:notes: :eye: :headphones: :eye: :notes:@bitracer @icekill

From the HD30 User Manual:

USB Mode Setting:

The USB mode selector switch on the rear panel determines the mode setting of the USB audio interface. Please note that the USB connector must be disconnected before changing this setting.

With the switch set to “A” (default), the USB audio interface is set to plug & play mode, and up to 96kHz and 24bits resolution can be achieved.

With the switch set to “B”, the USB audio interface can achieve up to 192kHz and 24bits resolution, and native DSD64 and DSD128. B mode is plug & play when using MAC, with the exception of DSD. With DSD you need additional software, please consult the HD30 section of the Hegel website.

I believe the native DSD support may be referring to i2s via AES/EBU input…

No, it just converts it to PCM and plays it.

:notes: :eye: :headphones: :eye: :notes:@icekill

This is the AKM DAC chipset… the C-Media device is the USB receiver chip…

:notes: :eye: :headphones: :eye: :notes:@bitracer

Maybe in this instance, however the C-Media chip supports i2s and this is most likely via the AES/EBU port. My feeling is the C-Media controller is causing the problem as it is the master bus controller as implemented in this design… it seems…

Thus may be a Belcanto RefLink would be a solution.
It could avoid the USB port input of the DAC and the CMedia controller.

:notes: :eye: :headphones: :eye: :notes:@bitracer

Yes there is something wrong with this… It’s not DSD signal… It is decimated DSD signal and a pale iteration of the source DSD signal.

:notes: :eye: :headphones: :eye: :notes:@icekill

The C-Media USB receiver/controller is the USB input to the AKM DAC… i2s is the direct link to the AKM chip… Perhaps the Ethernet connection would bypass the C-Media controller… You should consult with Hegel about your problem…

Don‘t think so. There is only one DAC that can do DSD over AES/EBU but it has FPGA based receiver.

It‘s not decimated, it’s mathematical conversion.

Sure it’s better to have DSD capable DAC if you want to play DSD files, but if you don’t, you do what you have to do.

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Yes it is a mathematical conversion that decimates the megahertz 1-bit Delta-Sigma signal to arrive at the LPCMxxx kilohertz word signal…

I would not snivel at a down-conversion to 705.6kHz… but from my personal experience 192kHz is a very pale iteration of even a DSD64 product.

You may be right… The C-Media chip overview describes i2s capability… We would need clarification from Hegel…

It’s downsampling, not decimation.

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Yes the result is a LPCM iteration of 24-bit words…
What do you think is happening to the DSD 1-bit sample-values that are not used in the LPCM code? …Based on the conversion algorithm design shaping the LPCM code, the unused 1-bit sample values are thrown-out…

I‘m assuming that Audirvana uses SoX for DSD to PCM conversation. I‘m sure that a brief Google search will give you plenty results on how it‘s done.

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When SoX is down-converting or better described as demodulating DSDxxx to PCMxxx, it is acting as software implementation of a Delta Sigma Demodulator and “the essential parts are generally a digital decimation filter - and optionally data refinement…”

*From the white-paper of the Infineon XMC4000 32-bit Microcontroller Series
Delta Sigma Demodulator (DSD) AP32302:

Delta Sigma Demodulator

The Delta Sigma Demodulator (DSD) task is to extract the analog information out of a bitstream. For this purpose the essential parts are generally a digital decimation filter - and optionally data refinement.

These type of filters are defined by the cascade stage (k) and the decimation factor (N). These factors influence the filter characteristics: response time, resolution, output period and filter group delay.

My understanding is that SoX first converts it to DXD, and then it’s resampled to whatever the target resolution is.

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Come to really think about what is going-on in Audirvana… I doubt the extra step is required, when the system knows the DAC sample-rate capabilities… SoX and r8Brain are PCM up-converters and PCM remodulators to DSDxxx… more likely than not the Delta Sigma Demodulation component is part of the Audirvana architecture… we would need clarification on this…

I demodulate DSDxxx to DXD 352.8kHz, offline in Bit Perfect’s “DSDMaster” application so I can employ 112db’s Redline Monitor HTRF Audio Units plug-in… DSDMaster’s only function is to demodulate DSDxx to whatever PCMxxx sample-rate and file type I wish… typically for me, it is 24bit/352.8kHz AIFF, The demodulation processing is actually fairly fast on a single DSD256 file. So…. I can believe, in Audirvana, the demodulation processing is done prior to the buffer and the plug-in component and the SoX or r8Brain up-sampling process in ‘real-time’… My best guess is the plug-in component and the up-sampling component is bypassed in the demodulation of DSDxxx to PCMxxx scenario… we will need clarification on this.,

:notes: :eye: :headphones: :eye: :notes:

My bet is it supports raw native i2s DSD via the BNC connectors… Even though the C-Media chip is the CM6632A and not the CM6635 that I posted previously here…

The Roon post confirms my previous statements regarding the DoP 1.1 PCM 176.4 kHz sample-rate limitation, if in fact the HD30 can play DSD64 files…

I doubt it. At best it can play DSD64, but ultimately it’s a PCM only DAC. Hence, I would just output PCM using the DSD to PCM DSD playback option in Audirvana.

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That’s quite a compromise in sound-quality from my stand-point… Give me a DSD64 playback everytime, if I have a choice between a demodulated PCM 24/176.4kHz iteration (the proper demodulation target) and the DSD64 source… the difference is stark in the subjective comparison…

This is not true… the AKM4490 chipset is DSD capable…

This all leads me back to seeing the CM-Media Speaker receiver/controller chip being the problem…