[433115]

I'm just wondering what the best settings are, considering it's capable of 32-bit 192k.
 
In Windows, I set the Default Sampling Rate to it's max, being 32-bit 192Khz.
 
I keep the sampling rate at "Same as Input" because I figure that's best, forcing something to be converted is a bad idea I figure.
 
And I would do the same with Format, but unfortunately this can cause an error because it's NOT capable of 32-bit IEEE Float apparently, and often times I get the error that "The Device isn't compatible with output format: 32-bit IEEE Float".
 
So, what would be better for me to set the Format to then? 24-bit int, 24-bit padded to 32, or 32-bit int?
 
Overall, here's my audio setup. Note that I only turn on the LAV Audio stuff when playback doesn't work without it (sometimes I need it for listening to things with too many channels etc). Let me know if you see anything that could be improved...
 
Sorry to ask but I'm just pretty... ya know... rookie or intermediate.
 
https://i.imgur.com/C9ideYs.png
 
 
Thanks!
 
- Alex

 

[KyPeN]

Maybe I'm wrong, but my understanding is that you should just set output to exactly what your music is and just forget about the rest.  So, it's probably just 16-bit + 44.1khz.  Upsampling doesn't add any information that doesn't exist and just burns CPU cycles.

 

[433115]

Right, but since when I leave the output as "same as default" it returns an error saying IEEE Float isn't supported.
 
That tells me that, in that case, the input must be 32-bit IEEE Float.
 
Sampling rate is left the same without problems, it's format that I have to change. I just wonder what it's best to "downgrade" the format to is, considering I'd like to have some somewhat-universal settings that I don't have to run into and tweak every time an error gets thrown.
 
Not really sure how it all works. Most audio I have in videos I listen to is AAC LC @ 44100 or 48000Hz, or FLAC @ some other rate (like 96000Hz), and most music I listen to is FLAC or ALAC, that's all I really know.
 
But when I play them, yeah, if I don't have a value chosen and leave it as "same as input" then it throws the IEEE Float error. Which means yeah, I've gotta define something.

 

[RRod]

  Right, but since when I leave the output as "same as default" it returns an error saying IEEE Float isn't supported.
 
That tells me that, in that case, the input must be 32-bit IEEE Float.
 
Sampling rate is left the same without problems, it's format that I have to change. I just wonder what it's best to "downgrade" the format to is, considering I'd like to have some somewhat-universal settings that I don't have to run into and tweak every time an error gets thrown.
 
Not really sure how it all works. Most audio I have in videos I listen to is AAC LC @ 44100 or 48000Hz, or FLAC @ some other rate (like 96000Hz), and most music I listen to is FLAC or ALAC, that's all I really know.
 
But when I play them, yeah, if I don't have a value chosen and leave it as "same as input" then it throws the IEEE Float error. Which means yeah, I've gotta define something.

 
There should be no issues leaving the sampling rate alone and changing the bit depth to an integer format > 16 bits. All that will happen is that the bits will get padded with zeros as necessary, which at least for me on Linux takes insignificant resources. On Linux you can poll the card for the available formats; my E-MU, for instance, only accepts 24LE so that's what I set things to.

 

[433115]

So essentially, assuming the source is 32 IEEE Float, it doesn't matter if it's changed to 24-bit padded to 32 or 32-bit int, just pick one?
 
Because apparently a lot of what I listen to is the float. Otherwise I would assume I wouldn't get this error thrown when I'm trying to play audio.
 
In my mind I was kinda thinking "Welp, guess I have to downscale it then, what's best to downscale to in order to cause the least distortion from conversion?"
 
I just assumed 32-bit int since that was the highest I could set it to. But I've read some places that anything above 24 is bad or... something.

 

[KyPeN]

Looks like you're on Windows.  What happens if you remove ReClock/KCP (whatever that is)/Oppo's software from the loop and only use Foobar2000 w/ WSAPI (no plugins), and your sound card set to 44.1khz 16-bit in the control panel?  You shouldn't need any of those things to get bit perfect output...

 

[RRod]

  So essentially, assuming the source is 32 IEEE Float, it doesn't matter if it's changed to 24-bit padded to 32 or 32-bit int, just pick one?
 
Because apparently a lot of what I listen to is the float. Otherwise I would assume I wouldn't get this error thrown when I'm trying to play audio.
 
In my mind I was kinda thinking "Welp, guess I have to downscale it then, what's best to downscale to in order to cause the least distortion from conversion?"
 
I just assumed 32-bit int since that was the highest I could set it to. But I've read some places that anything above 24 is bad or... something.

 
Just pick one. Conversion from float to int would just be multiplication followed by truncation/rounding, which at these bit depths wouldn't possibly be audible. I have no idea why 32-bit int would be "bad" compared to 24-bit int. There isn't anything that gets 24bits of actual dynamic range anyway, so good/bad is highly theoretical at these numbers.