I want to thank everyone for replying, and now I will attempt a reply to things still leaving me curious. As if my first post wasn't enough.. >_<
The limits of audio reproduction today is 20 bits/120 dB or so. It can be shown that 20 bits/120 dB essentially captures the full dynamic range of a live concert. Beyond that, both technologically and practically, there is no point. So 20 bits is perfection.
Is this due to microphones used in the recording industry?
I just noticed in the latest update of Windows the sound control panel has disappeared! And with it, the above option. It is a pain now to figure out what the heck is going on.
Yeah, you can only imagine how lost I am with rudimentary knowledge on most of this stuff to begin with on a theoretical level..
You can also use software to upsample by using Windows to do it (not very high quality) or use the one in your player (e.g. Roon). My advice: don't bother. Play content at its native rate by using ASIO or WASAPI.
What about when I want to use NOS for the purported "better impulse response"? Would that be an okay exception? Also, why can't DAC's be forced to perform the upsampling when using ASIO/WASAPI? Instead if I want to upsample, I have to leave it to Windows (since I use foobar, I suppose I can use an addon that is supposedly a "better upsampler" than Windows?
Some people like it and record music in it. If so, I prefer to play the DSD version as opposed to whatever they did to convert it to PCM.
I understand everyone has a preference/opinion on this somehow. What I don't understand is why, and how articles that argue one or the other can be dismissed? Also I have DSD capability, and I've tried it (on the RME if you don't want the PCM conversion to be able to control volume on the DAC/OS, you can only output directly to Line-Out with no volume control). Now here's why I absolutely can't understand DSD.. it clips and there isn't anything I can do about it. So to me, DSD is worthless from a practical standpoint. And as the article I linked says, there aren't any serious widespread true native DSD editing software, so those "true DSD" recordings seem practically unwieldly in the real world.
And no .. impulses aren't reproduced better. What one hears is poor reproduction with a sharpish/rawish 'edge' combined with some 'roll-off'.
NOS aficionados just like/prefer this presentation. Better it is not.
When upsampling is used (say 4 or 8 times) this effect is as good as gone, steps become smaller and higher in frequency.
The obligated post filtering that would be essential for NOS 44.1 files is taken care off by the filter in the upsampling algo.
I heard no rawish edge, unless that edge is like a knife being rapidly sharpened and the sound effect is being warped electronically. I
read this thread recently and MC seemed to say that impulse response is improved, so I think I need to do some reading because perhaps I am thinking of the wrong aspect or under the wrong condition. So would you say NOS is pointless in 192kHz as well seeing as the "ringing" is gone, and if that's gone, that's what you were looking for, while this whole impulse response improvement talk is nonsense?
All DS DACs upsample, its how they work. I don't think it is a problem to do it in a DAC.
In software with a fast processor one can probably achieve slightly better technical results. I'll lay of the audibility of this.
Miska's player can upsample (and does a good job at it but is a horrible player to use). There is a trial period. Play with it.
For R2R DAC's things differ. IMO it is good practice to upsample before sending it to the DAC. Certainly when the DAC's have no or poor post filtering.
So in non-R2R DACs, upsampling is 100% pointless? But if the DAC's themselves do it, then why would the functionality exist in the first place. Also why don't DAC's have some setting where you can force them to do the upsampling (or is this non-rational, because they already do it, and even if they didn't upsampling does nothing to begin with)?
This may sound obvious, but I am guilty of neglecting it when I first installed EQ APO and getting distorted sound:
In the lower left panel of APO, make sure your peak gain is not above zero (see pic below). If it is, reduce "gain" in "preamplification".
Yeah mine was set to this. The ringing didn't cease, and I also use -10dB pre-amp when testing EQ settings for HD6XX's (I'm in the camp of reducing frequencies when EQ'ing as much as I can). So I don't think this was the issue in my specific case.
Right. What I forgot to add was that it's often a good idea to set the software volume control to slightly below 0 dB. For example -3 dB. This mitigates the potential for intersample peaks which most DACs do not handle well.
Yes, I take clipping serious the moment I got into audio, and never have dB at FS from my DAC, or if the DAC didn't have volume control, then Windows system audio was lowered. Also you're 100% right, even -1db isn't good enough, the latest update to the RME DAC exposes output signal power (and some modern music just yesterday I was playing was clipping even at -2db, I couldn't believe it.) Oh and I really appreciate your explination with driver-level ASIO/WASAPI and such, I've read your posts with others where you dismantled "DirectKernel" vs "ASIO" and such. Very enlightening, and thank you very much for that. I will reply to you lengthy post after this one, since mine is getting massive. Though I wanted to say, for some reason, I see the actual audio content IS clipping, but not my out-put audio (nor do I hear it clipping in reality). I guess that has to do with the following posts where the guy talks something about some DAC's handling this issue better for some reason (I need to learn the differences between all that "dB" denominations it seems).
