16 bit files almost unlistenable now...

[audiventory]

14 hours ago, Cary said:

Noise floor of 50db in your room + a 90db dynamic range for 16 bit = 140db peak levels to exploit the full dynamic range.

 

Dynamic range is not definded as difference between noise floor and peak level.

It definded difference between levels with allowable distortions.

Read details here https://samplerateconverter.com/educational/dynamic-range

 

I think, when acoustical noise level (air pressure) of record on 50 dB, it's too much, even worse than analog sources. I suppose, 50 dB of the acoustical pressure is like to cooling fans noise at hard working notebook.

 

Probably, noise floor may be from 0...10 dB in the listening room.

[audiventory]

2 hours ago, GUTB said:

High resolution audio sounds better, that's a ** FACT **.

 

In high resolution, It is easier to implement lesser distortions technically.

 

For me 44.1 kHz and DSD64 are most sophisticated in implementration.

[audiventory]

6 minutes ago, Summit said:

The quality of the recording is the most important,

 

ADC distortions also depend on higher sample rate.

 

For some kinds of editing sample rate is not big matter though.

[audiventory]

9 hours ago, jabbr said:

If you mean by "aggressive" as "processor intensive" then yes, indeed processor intensive. 

 

Noise shaping is defined by filter in feedback of a sigma-delta modulator. See "DAC" in feedback at the picture "Sigma delta modulator inside" here https://samplerateconverter.com/educational/how-work-sigma-delta-modulation-audio

 

Remark: there are number of different modulator schemes, but in the post described general principle.

 

More shaping steepness (filter frequency-magnitude steepness) > more sensitive feedback > more unstable system with feedback (modulator).

 

Steepness of the shaping depend on the order and design method of the filter.

1 order = 1 [multiply+sum] operation.

 

As rule, sigma-delta modulators have orders lesser 10. And changing of the shaping steepness don't make big performance difference.

 

Resampling filters and big sample rates (big number of calculations) consume most of performance.

 

[audiventory]

29 minutes ago, Andyman said:

Here is a chart showing quality in one simple graphic.

 

To estimate quality, it is necessary to define measurable value (distortions, method of audio perception testing, etc.).

[audiventory]

7 minutes ago, Andyman said:

I didn't post the graph but was questioning it

 

Looks like I done something wrong when quote.

[audiventory]

7 hours ago, GUTB said:

You can only do basic mixing/editing on these without destroying the benefit of DSD.

 

By available publications, editing 1 bit DSD to 1 bit DSD without re-modulation here is cut/merge only.

 

 

Mixing 1 bit DSD to 1 bit DSD is impossible without re-modulation.

 

Without filtering, part of dynamic range is lost. Filtering give ringing.

 

However, mixing (both PCM and DSD) can use digital filters and have ringing and amplitude/phase distortions. It may become matter comparing with filtering DSD.

 

Analog mixing have non-linear and frequency distortions, noise.

 

When record is played back PCM converted to DSD with ringing during upsampling.

 

Pure PCM R2R DAC have more issues, than DSD modulated.

 

Resume:

If record is edited, I don't see problem in accurate conversion DSD to PCM for editing comparing other issues in given context.

 

 

 

 

[audiventory]

19 minutes ago, adamdea said:

I do however feel confident you will struggle to hear 16 bit quantisation noise

 

In my opinion, audible or inaudible noise is not matter here: signal/noise ratio make sound quality.

 

Example:

 

[We have white paper. White font is invisible. Black one is visible.]

 

We can replace "white" to "yellow" or "blue" or other. It don't change change of 3 sentences above.

[audiventory]

51 minutes ago, Jud said:

I bought a "lifetime" subscription

 

I can't understand how to fund development, researching, make product better and support it with "lifetime" subscription.