DSD Frustrations With Manufacturers

[Ryan Berry]

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21 hours ago, The Computer Audiophile said:

Just testing out the Ayre QB-9 Twenty and found the DAC supports DSD256 at 12.288 MHz (multiples of 48 kHz). However, the front panel display goes blank with this sample rate. Playing DSD256 at 11.2896 MHz, the AB-9 Twenty front panel correctly says 256.

 

@Ryan Berry, can you weigh-in on this one? Is everything playing correctly at both 44.1 and 48 multiples, but just the display doesn't recognize DSD256 at 12.288 MHz?

 

Hey Chris,

 

The QB-9 has handlers written for DSD256 at 11.2896 MHz but not 12.288MHz right now.  We looked into this before releasing the QB-9 and could not find any source material that was recorded in a DSD format with a multiple of 48kHz, so it came off as more of a marketing gimmick to us than actually a useful feature and only adds confusion to what DSD256 is.  Adding a handler is REALLY trivial, as I imagine it would be for most DAC manufacturers.  The only hesitation is if there's any actual gain doing so.  If there's some source material out there we're missing, let me know and I'll talk with Ariel some more about it. 

[Ryan Berry]

49 minutes ago, The Computer Audiophile said:

Thanks Ryan.

 

This is more about using apps like HQPlayer to upsample content, using different filters and modulators, to DSD256 and keeping the sample rate as a multiple of the original. I assume the QB-9 upsamples internally to a multiple of the base rate. This is very similar just done outside the DAC and enables the user to select what he prefers sonically.

 

Hey Chris, I see.  To be honest, we've never been big proponents of upsampling outside of the FPGA, where we can do everything in a single pass to minimize rounding errors as well as ensure that the original data is preserved and not rounded out by some software post-process.  We only can guarantee what WE do internally, so this gives us a bit more control on making sure the end product is as true to the original as possible.  I'm sure you already know how we tend to think of DSD based on posts from Charley over the years, but I still think that converting a 192kHz sample to DSD is a fair argument for a 48kHz-based DSD rate being available if you're really set on doing it.  Let me discuss that aspect with Ariel.  It's just a matter of adding a line to the code and figuring out if there's a good way to display what it's doing with the characters available on the display.  

 

41 minutes ago, jabbr said:

I gather than when converting from say 24/96 PCM to say 44.1x256 DSD that certain filters need less processing when staying within rate family, yet I have not heard that the final result depends on staying within rate family. For chip families such as ESS which operate internally at ?100Mhz we aren't staying with a multiple of either 44.1 nor 48Khz regardless.

 

Well, it CAN operate at 100MHz.  I wouldn't ever recommend using it that way.

[Ryan Berry]

Just now, The Computer Audiophile said:

Thanks for the clear reply. I certainly hear you Ryan. 

 

I love the ability to make my own choice, in essence to have control over what I do when I want to do it, and to turn control over to you when I want. Thus, using HQPlayer when I want to and using the Ayre FPGA when I want. Choice is wonderful :~)

 

I highly recommend checking out @Miska's HQPlayer. I won't speak for him but I'm guessing that anything that's done in an FPGA can also be done with his software. The cool thing is that its got endless power because it runs on full blown PCs. 

 

I'd say leaving the display at 256 no matter if the playback is based on 44.1 or 48k is cool. 

 

Thanks again!

 

 

 

 

I know HQPlayer pretty well, we've used it in the past for testing and agree that it's a nice piece of software.  However, you will still run into a second pass of oversampling being done at the FPGA to get to 16X and apply our Minimum Phase filter, so you'll run into what I mentioned before.  

[Ryan Berry]

Just now, jabbr said:

 

Thanks, I don't claim to know how the ESS chips work internally. Regarding DSD @ 48k multiples, I'm really just trying to understand if there is a significant benefit with 48k x based PCM source conversion. I can do it either way on my system. My brief testing seemed to indicate that there was no benefit for me regsrding "SQ" ... but I am interested in learning.

 

That's a big topic full of a lot of opinions.  Some groups studied could tell no difference while a study in Tokyo suggested people could hear the differences between DSD and PCM with a preference toward DSD (though no differences heard between different rates of DSD, interestingly).  I've not heard much benefit of doing a conversion to DSD myself and I know that Charley really didn't care for the format, but that was probably more on principle than anything.  Either way, I tell people to trust their own ears and just enjoy the music.  The argument on what is right or wrong would go on endlessly and we'd all be sitting here debating on forums instead of listening to music.

[Ryan Berry]

23 minutes ago, Miska said:

 

With DSD inputs too? Or does DSD inputs bypass your processing and pass through to the DAC chip?

 

 

That's a fair point that I wasn't thinking about in DSD terms, Miska.  Good catch.  DSD passes directly through to the ESS untouched, so it wouldn't be subject to another round of oversampling like pre-upsampling PCM would.  It doesn't eliminate some of our other issues with the format as a whole, but you're right that it wouldn't be subject to double-oversampling.

[Ryan Berry]

52 minutes ago, bobflood said:

Interesting studies. I would love to know if for the PCM playback multi-bit hardware was used. For true PCM rendering, the DAC must be able to process a 24 bit word and assign it a specific impulse value. R2R ladder DACs are the classic example but there are others. If PCM material is fed to any one bit chip DAC, it first has to be converted to a bitstream (DSD) and then modulated. So what one is really hearing is the chip manufacturers version of DSD playback not true PCM playback. Any difference heard when using a one bit chip DAC for both PCM and DSD material is by definition likely to be slight.

 

The more I read, the more questions I end up with.  Are the files being tested conversions of a PCM file or rerecorded with a DSD ADC?  What if the ADC being used for PCM had a filter like the QA-9 instead of a brick wall filter?  How is the DSD vs. PCM data handled inside the DAC?  The list goes on and on and every test is likely flawed in some way.

[Ryan Berry]

Just now, The Computer Audiophile said:

What I wouldn’t give for an entertaining Charley rant on how DSD came to be, as I sit in my quarantined listening room :~)

 

I can at least give you an old one!  http://www.the-ear.net/features/dsd-under-fire