The tradeoff between DSP and DSD

[Miska]

5 hours ago, Ron Scubadiver said:

If you want DSP, my research and informal listening tests indicate a high quality conversion to PCM without converting back to DSD is the preferred approach.  This is possible in both JRiver and Foobar2000.   It helps if your DAC can input a high bit rate PCM stream, like 352k.

 

...and then your DAC chip will do equivalent thing inside (conversion from PCM to DSD-like data), in a not so great way...

[Miska]

2 hours ago, mansr said:

Room correction works up to 50 kHz or so. Speakers and microphones only extend that far, if you're lucky, so for higher frequencies you have no measurements on which to base a correction. For these purposes, even 96 kHz is thus sufficient, and I don't believe resampling to this rate around the correction filters will be of detriment.

 

I usually measure up to 20 kHz and then have rest of the response just flat-out to what ever the Nyquist in question happens to be. So I run the room correction always at the source material rate. So no need to have rate conversions for room correction purposes.

 

However, as usual, I use upsampling. Since highest rate DSD material I have is DSD256 and many of my DACs support DSD512, the output is anyway DSD512. Provides also some extra headroom compared to the original material.

[Miska]

1 hour ago, mansr said:

Would it make sense to lowpass somewhat below the tweeter breakup frequency?

 

Yes, certainly it would. I just personally avoid using those peaking dome types that need it...

 

1 hour ago, mansr said:

There's no need if your computer is fast enough. Depending on the complexity of the correction, performing it at a lower rate can be cheaper even with the overhead of resampling.

 

Sure!