on the idea of upsampling/oversampling low resolution files, it isn't always advertised, but pretty much all modern DACs do it without telling you already. so really just buy something convenient that seems to go with your budget and don't think too much about it.
Or alternatively, think about it a lot, and take a class in informatics/information theory.
It's a fascinating field, and applies to a lot more than just sound reproduction. I just wish I had the mathematical nous to understand it better.
hehe, sure I'm not saying it isn't interresting or vastly important as a process(that I also fail to fully understand). I'm more talking about how when sound can benefit from upsampling/oversampling it usually is already done anyway as NOS DACs are a thing of the past(or should be ^_^).
That's my impression too. I always thought PCM was just a raw bit stream that hadn't been packaged into a codec yet. I have no experience with DSD, but can't imagine how you could re-package PCM bit stream to improve the quality. It should be the same exact bit formation, any extra data is just filler, and any less would require a codec. I never understood why PCM was better than a packaged codec anyway. Sounds the same to me once the DAC gets a hold of either one. The only major benefit to PCM I found is universal compatibility. With one foot in PC systems and another foot in Mac systems, I take advantage of that quite a bit.
well from what I understand, some DACs chips(the real D to A part) speak PCM, some speak DSD. so for foreign languages like AAC MP3 FLAC... it's only a matter of who's doing the transaltion ^_^. up till now all of those codecs were built with PCM in mind so it's obvious that they're more comfy with it. but it wouldn't be that hard I guess to get a new codec that speaks DSD as second language. it's just not what we use now.
so at the moment if I use MP3, it will turn into PCM anyway and deciding to output it to DSD would force a double translation. even if it doesn't end up with loss of signal quality, it looks to me like a waste to do that for no concrete benefit.
/!\ warning I have no idea if any of the following is true, I didn't actually find or understood enough to make sure of it all, it's just me going wild. I'd be happy if someone could either confirm, or show me the light
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but for anything working with pulse modulated signals I couldn't really understand how it really was different for PCM and DSD. the little I understood was that for PCM the DAC gets a sample with a value equivalent to a voltage value, and then does whatever it has to do to get to the right voltage using the appropriate voltage impulses. for DSD the signal already give the timings for the impulses, but the result sure look the same.
then I thought I had it when I realized that DSD DACs were one bit and PCM DACs where 24 or 32bit. but even that doesn't seem to stand as there are DSD using several bits(I'm guessing to reduce the noise from using a stupid 1bit signal). and there also seem to be some PCM dacs that actually use less bits as they don't really need a lot for pulse modulated signals.
so the more I look into it, the more it seems to me like DSD is just a rip off of delta sigma that pretends to be different by writting down time values instead of amplitude. but sound being made of sine waves, both are always linked directly and say exactly the same thing.
/!\ end of warning!