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Patched foo_input_sacd version for high-quality DSD->PCM (88.2/96 k
amiti
post Apr 4 2013, 17:10
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QUOTE (audiogene @ Apr 4 2013, 18:54) *
Aside from your question which I cannot answer either the patched plugin is not working in recent foobar versions. I tried it out because I wanted to carry out some research about the plugin's quality but foobar refused to install it.


I am using foobar 1.1.18. Since which version you have troubles with this plugin?

I am guessing that if auto install doesn't work anymore you can install it manually as in the "old days".
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marc2003
post Apr 4 2013, 17:30
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i'm guessing it's because the files are inside a subfolder in the zip. if you move them to the root, it should work.
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db1989
post Apr 4 2013, 17:57
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QUOTE (audiogene @ Apr 4 2013, 16:54) *
I am interested in answers about the best sounding cut-off filters - and/or the theoretically best ones. And is there a single answer to the question if double precision or floating point math is sounding better (and again: should be theoretically better)?
In line with #8 of the Terms of Service, members shall confine all discussion to mathematical measures, and reports of supposed audible effects are not welcome unless they are objectively demonstrated.
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tedsmith
post Apr 4 2013, 19:01
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QUOTE (audiogene @ Apr 4 2013, 08:54) *
Aside from your question which I cannot answer either the patched plugin is not working in recent foobar versions. I tried it out because I wanted to carry out some research about the plugin's quality but foobar refused to install it.

Does anybody have some advice which is the proper (most "audiophile") setup for the offline conversion of DSD->PCM with the original foo_input_sacd plugin (not taking into account CPU load).

I am not interested in an answer regarding the (re-)sample rate (which I will chose dependent on the highest *usable* frequencies in the DSD material, which oftentimes does not go beyond 22kHz). I am interested in answers about the best sounding cut-off filters - and/or the theoretically best ones. And is there a single answer to the question if double precision or floating point math is sounding better (and again: should be theoretically better)?

Thanks for all contributions - theoretical and practical ones wink.gif

/audiogene

Nothing is that simple.

All digital filters have a tradeoff between steepness of cutoff and "steepness of time". If you insist on a very steep cutoff you'll smear the signal in time (preringing, etc.) If you insist in a perfect impulse response (i.e. total time alignment) you'll have a shallower frequency response cutoff. So pick your poison. Thank goodness that most of the filtering necessary in sigma delta conversions (loosely all but upsampling from 44.1 or 48k) have room for a more relaxed cutoff and you have more freedom of choice.

Perhaps swamping all of the above you just have to be careful with your math. No particular selection of double vs single precision floating point or any fixed number of fixed point bits is a panacea. The worst thing to do is to make some assumption like "if I always use double precision floating point I'll never loose info...", that's the mindset that leads to the most errors. All of that said, the filtering needed to go from 1 bit DSD to 24 bit PCM can be done well with careful single precision floating point or more easily with double precision. But personally I'd choose careful fixed point: the math to prove numerical accuracy is much simpler for fixed point filters.

You'll have to judge the sound differences (if any) for yourself. Personally I write all of my own upsampling and downsampling and have a preference for IIR filters but they have their own math problems smile.gif

Adobe Audition 3.0's lowpass scientific filter set for a > 6th order Bessel response with a corner at 50k to 80k is reasonable for an offline conversion of 1 bit DSD to 2.8224MHz PCM. Then you can use your favorite sample rate converter (including Audition's) to finish the job. In a pinch I've gotten fine results doing just that tho I'm pretty sure no-one does it that way.

No particular spec is useful for comparing DSP algos: Unless you do it yourself you're at the mercy of the skill of the coder of your DSP...

This post has been edited by tedsmith: Apr 4 2013, 19:17
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