DSD Frustrations With Manufacturers

[Miska]

Funny thing is that they have clocks already for the 48k-family PCM playback. They just fail to switch to that clock with DSD input.

 

Even worse are DACs that have lost this 48k-family capability due to firmware update (iFi, when they introduced MQA support).

 

[Miska]

Good side is that most of the poly-sinc group of filters in HQPlayer can convert from 48k-base to 44.1k-base output. But it still limits available filter choices, because it cannot be done with all filters.

 

[Miska]

5 minutes ago, One and a half said:

Please pardon my ignorance,  which DSD source material is based on 48kHz multiples?

 

Don't seem to be offered 3.072MHz music from Blue Coast, Analog Productions, even the pirates... ☠

 

You can easily create such. I can also record such with my two RME ADI-2 Pro's. Up to 12.288 MHz.

 

[Miska]

2 hours ago, jabbr said:

Is there any need/benefit to 48x256 (or any of the 48 rate family x DSD)?

 

Yes, if you want any filters that can do only integer or power-of-two multiples...

 

With single stage poly-sinc filters in many cases it is is also lighter CPU load to work within rate family. For example try poly-sinc-xtr-lp from 48k to 48k x512 or alternatively from 48k to 44.1k x512. Latter works at least on RTX2080Ti, but not on i9-9900KS while the former I think does.

 

[Miska]

10 minutes ago, Ryan Berry said:

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.

 

HQPlayer can do 1024x upsampling in a single pass, and on top of that run digital room correction filters and such with millions of taps.

 

Nice thing also is that since software processing runs asynchronously from any sample clocks, it can monitor the output and and also upcoming future input data, and re-process data based on decisions, while still meeting delivery deadlines. This is possible because modern CPUs can run at 5 GHz clock speeds. And modern GPUs, like my Nvidia RTX2080Ti with it's 18.6 billion transistors can do massive amount of DSP operations as well. All at very reasonable cost.

 

[Miska]

9 minutes ago, AudioDoctor said:

Chris, you could always get a DAC that doesn't do any upsampling, or oversampling, like, say, a Lampizator Pacific... 😉

 

Or maybe one of the Holo Audio or Denafrips devices?

 

[Miska]

47 minutes ago, Ryan Berry said:

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.  

 

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

 

[Miska]

4 minutes ago, jabbr said:

Being sure that I am interpreting this correctly ... if software processing runs asynchronously from any sample clocks, then what is the dependence of the final output rate family on the input rate family? (aside from some additional processing which is offset by a slightly different output rate)

 

Dependency is mathematical. Processing is clocked by the CPU and GPU clocks which are not related to input or output clocks.

 

This is totally different from typical synchronous process you have in a DAC for example.

 

[Miska]

30 minutes ago, The Computer Audiophile said:

An easy way that I’ve always looked at the sample rate family issue is to consider how it works when downsampling. 
 

A recording done at 88.2 can easily be made ready for CD at 44.1 by removing every other sample. No math involved. 

 

Without proper decimation filter that would give you plenty of aliasing, as all the content between 22.05 and 44.1 kHz would appear in the 0 - 22.05 kHz band with inverse spectrum.

 

Same way, you could upsample by copying every sample twice, but without proper interpolation filter this would again produce images of content from 0 - 22.05 kHz band between 22.05 and 44.1 kHz again with inverse spectrum. Note that most DAC chips use this method at and above 8x rates.