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The tradeoff between DSP and DSD


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21 minutes ago, sdolezalek said:

What part of "downsample it by 8X" am I misunderstanding? 

That's from the description of how they convert DSD to PCM. He then goes on to say that this step is skipped for DSD to DSD processing.

 

The main effect of doing DSP at DSD rates is vastly increased CPU load. DXD or double DXD rate provides ample margin for filters between the highest music frequencies and Nyquist.

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1 hour ago, firedog said:

But I guess some of you understand it better than Michal, Thorsten, and Brian.

They probably understand it just fine. They just choose to misrepresent things when it suits their commercial interests.

 

1 hour ago, firedog said:

You should design and sell your own HW and software. 

I guess giving it away for free isn't good enough.

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6 minutes ago, sdolezalek said:

I have a ton of respect for Brian's work and so I'm not surprised that he figured out a way to do DSD equalization that is not destructive.

It doesn't exactly take a genius to skip the resampling steps and do whatever DSP at the higher rate.

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3 minutes ago, sdolezalek said:

On a more serious note, if this is so easy, are you suggesting that Dirac, REW, Acourate and others providing DSP corrections are doing the same to preserve the integrity of DSD input signals?

I don't know what they do, or even if they support DSD at all.

 

3 minutes ago, sdolezalek said:

If not, are you of the view that the benefits of their corrections clearly outweigh any damage they may be doing to the original sound wave?

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. If you worry that the resampling filters will have and audible effect, you can pick a higher rate like 384 kHz and still reduce the CPU load compared to running the main DSP at DSD rates. A clear benefit of using a lowish intermediate rate is that it enables the use of a more sophisticated correction filter without exceeding available CPU resources.

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3 minutes ago, Samuel T Cogley said:

I, for one, would like to better understand this "widening from 1 bit per sample to 64 bits per sample" business.  I've heard of multibit SDM, but I'm not really understanding how this works practically.

It's quite simple. For each DSD input bit you create a sample of whatever width you like, such as 64 bits, with the most positive or most negative value used by the data type depending on whether the input was a 1 or a 0. Then you proceed as you would with any PCM input. To get back to a 1-bit DSD format, a sigma-delta modulator must be used.

 

Multi-bit SDM is something else. That is a sigma-delta modulator with more than two output levels. Even a small number, say 5, gives much more flexibility in the design of the noise shaping filter without running into stability issues.

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1 hour ago, Miska said:

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.

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

 

1 hour ago, Miska said:

So I run the room correction always at the source material rate. So no need to have rate conversions for room correction purposes.

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.

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