Tidal chucking MQA?

Are the recordings really native DSD? There is only one ADC, IIRC, which happens to be made by AKM, and which happens to operate in native DSD mode. Pretty much all other modern ADCs do not actually internally digitize to DSD nor to PCM, although they can be configured to output PCM or DSD following some internal DSP processing.

But who cares? If a particular dac sounds better playing back in DSD mode, then why not always send it DSD (even if the source material is CD)?

Also in case its not clear, the reason for using DSD for playback is IMHO mostly because some dacs sound better in DSD mode as versus PCM mode. Prior to being processed by a dac, its probably better if source material is in PCM format. Recording, mixing, and mastering is mostly easier to do and potentially more powerful if done in PCM.
It would be awesome if there was some PCM to DSD converter that converted everything in realtime, and that would actually make the kusic almost as good as native DSD. I suspect that it will not sound as good native DSD, or even close, but I have not tested it. Seems like a lot if hassle, but if it was possible then why not.

Regarding DAC quality, if $350 is what I have to spend on a DAC, then that DAC will sound good enough to me. :)
 
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I believe there are other devices that were available that recorded natively in DSD from at least Sony. I believe the only currently readily available devices are fro AKM. I have several devices capable of recording natively in DSD, and yes they are all based on AKM ADCs. I have an RME ADI-2 Pro FS R that supports native DSD as do my kaamos tech converters.

https://www.alpinesoft.co.uk/VinylStudio/dsd.aspx

My main system is build around PCM hardware running at 2496 so no DSD here, but I do transcode DSD to PCM for playback, my other two systems are analog with dacs that support DSD. I have native DSD and recordings transcoded from DXD.

I am coming to care more about the end result than the means of achieving it, or at least that is what I tell myself. :)
 
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It would be awesome if there was some PCM to DSD converter that converted everything in realtime, and that would actually make the kusic almost as good as native DSD. <snip>
Regarding DAC quality, if $350 is what I have to spend on a DAC, then that DAC will sound good enough to me. :)
Roon has it in their 64 bit DSP, I recommend you try it. A cheap Intel Nuc or similar computer is all you need to get started. It is not exactly "real time" as there is some latency which is processor and memory dependent. I think latency is irrelevant practically speaking. There are undoubtedly other options for doing the job, but Roon has a free trial option, if you do not like it cancel within 30 days.

My individual dac budget is about 2x - 3x that and I would still consider those inexpensive.
 
It would be awesome if there was some PCM to DSD converter that converted everything in realtime...
There is a very good FPGA freeware one in the forum (for diy use only): https://www.diyaudio.com/community/threads/simple-dsd-modulator-for-dsc2.370177/ (I use version 3 firmware)

And there are some accessories to go with it: https://www.diyaudio.com/community/threads/general-purpose-dac-clock-board.413001/ Note there is an interconnect board on page 4. Reclocker schematic is on page 3.

Combined with a TheWellAudio or with a MarcelvdG RTZ FIRDAC, you can go in a direction well past anything from Gustard or Topping. https://www.diyaudio.com/community/threads/return-to-zero-shift-register-firdac.379406/ ...The trick is configuration as shown in the Clock Board thread plus a transformer based output stage. ThorstenL hints at one way to do it.
 
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That's entirely possible.

On the flip side: Have you seen any hard evidence from MQA that their system works and does not degrade the original 16-bit data? I haven't. MQA is the one making the claims out audiophile superiority. They should be the ones backing up their claims with measurements.

Tom

Isn’t CD quality considered lossy to starting with, no?
Most likely original studio master @96k 24bit and MQA claim as promoted:

“ ability to preserve more of the original signal’s information at that level. This means you’ll be able to hear your favorite tracks in stunning detail, just as the artist intended (from the studio master)”

… that can be squeezed into CD bandwidth

So, Tom wrote something about degrading original 16 bit data and acko pointed out that the original recording probably isn't 16 bit anyway. No idea why this derailed into a completely off-topic discussion about analogue tape recorders and vinyl records.
 
The only bad aspect of the DSD "upsampling" is the exaggerated energy usage.
Its nothing compared to a class A/B power amp biased a little more into class A. The only energy needed for DSD upsampling is to power an FPGA that uses less power than a USB board. That said, there are less efficient ways to do the conversion too, just like there are inefficient fully class A power amps.
 
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I have never measured the power consumption of my dedicated Nuc based Roon core, but DSD conversion is definitively processor intensive - the Nuc flat out consumes about 60W which is far less than my four channels of 300B SE class A mids and tweeter amplification and class D amps (bass and subs)

The tape/vinyl analogy appeared due to some earlier comments Mark was responding to.

The widest dynamic range source material I have in my collection is 2-track 15ips master tape reissues, despite the obvious theoretical advantages 24 bit PCM in particular has in this area. IMO there is a point where dynamic range can become excessive, and that is when you set the volume to a comfortable level for quiet passages, and are then forced to turn the volume down on very loud passages due to auditory discomfort.
 
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So, Tom wrote something about degrading original 16 bit data and acko pointed out that the original recording probably isn't 16 bit anyway. No idea why this derailed into a completely off-topic discussion about analogue tape recorders and vinyl records.
In interpreted this post as "all digital is lossy; only analog is lossless". I simply pointed out the flaw in that argument. Analog formats are quite lossy, more lossy than current digital formats due to their much lower dynamic range.

My point was actually NOT that the original recording probably isn't 16 bits anyway. My point was that any analog format, be it reel-to-reel, compact cassette, LP, 7" single, wax cylinder, whatever, will have a lower dynamic range than even a 16-bit CD without any of the fancy dithering or noise shaping. I.e., any analog format will be lossier than even a 16-bit CD. So if you are looking for distribution and reproduction that is as true to the original as possible, i.e., as lossless as possible, the CD is the better format. Noise shaping such as MASH only further improves on this.

Of course, these days there's really no reason not to distribute the 24/192 recordings. Bandwidth is plentiful and storage is practically free. Modern DACs support 24+ bits. So I simply don't understand the point of MQA. It seems it's trying to solve a problem that isn't there all while creating another problem ... and charging users for the experience. Yeah... No thanks.

Tom
 

TNT

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Be sure that MQA was 95,34% a commercial endeavour disguised in some juicy technical lingo for the gullible to drool on... music had very little to do with it... thats why it was so devilish....

What the industry needs is something new that is demonstrably compleatly different and superior to what we hear today... anything else will just pass by...

This was for sure not M Q A.....

Listening / comparing MQA gave at hand a very small difference... which probably mostly depended on new mastering for the release rather than some "folding"...

Yes, I deplore it.. ;)

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TNT

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I cant see how any dependency level of anything could decrease that statement validity. Cant see that income has any bearing on this at all... Your sentences are not really coherent so har to say how you see the income aspect - i would say the higher gain/income, the more need for evidence....

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