flimflam wrote: > Same thing! Smoothing filter is a specific and recognised term for this > filter - whether you like it or not :-) ! The term is used by many, > Analog Devices refer to it by this name a lot. That the stair steps are > smoothed seems obvious, the term does not mean to imply anything else. > OK, may be. I learned all my signal theory in German so terminology might not be my strong point here.
But it's still something different from the interpolation filter we were talking about above. > > The latter is all about the former though, of course! It is the raison > d'etre, as you say yourself, behind oversampling. John was explaining > this in relation to stair steps of the ZOH but you did not seem to > understand and asked what stair steps, I was trying to explain, that's > all. > Um. Sorry. Again, I fail to see it. Again: Why does interpolation filtering on the digital side improve the analog filtering and how is the interpolation even correlated to your desired filter response. I can find a lot of prosa about this on the internet, mainly by makers of such filters, but I haven't found a single explanation that makes sense. All I find is stuff like this: http://www.ni.com/white-paper/5515/en/ Which - sorry - is nonsense (the digital interpolation part. The analog and imaging explanations are OK). I mean: it's not if your goal is to reproduce ideal sine waves generated by a signal generator as shown in the example, fine. But to do that you'd not have to take such an effort, perfectly reproducing sine waves from a few samples is the easiest thing you could ever do in signal processing. For a real world signal like music I fail to see how an interpolation filter will be effective and valid because - unlike in these sine wave examples - you do NOT know the exact waveform. The only reason the "interpolation filter" in that example is effective is because whoever drew that diagram did EXACTLY know the parameters of the sine wave PLUS, these parameters didn't change at all over the duration of the signal. Under these circumstances it's of course easy to fill in the added samples with perfectly fitting interpolations hence not creating any harmonics. However, that's NOT what you have in real life, ESPECIALLY not with a software like sox that can only look at a few samples at a time (you COULD theoretically do this right when you did an analysis of the whole audio file but we are talking about a LOT of digital signal processing here). > > I agree 100% with you. I would avoid external resampling, and deliver > bit-perfect to the DAC, preferring to let the extremely capable > designers at Analog, Cirrus Logic, TI, Wolfson etc do their job. If a > DAC/ADC loop cannot be distinguished from the original, it says a lot I > think. That's my opinion, it's not a very positive contribution to the > thread and its undoubtedly hard work, so I didn't say it loud before. Now... THAT's a statement I understand :) --- learn more about iPeng, the iPhone and iPad remote for the Squeezebox and *New: Logitech UE Smart Radio* as well as iPeng Party, the free Party-App, at penguinlovesmusic.com ------------------------------------------------------------------------ pippin's Profile: http://forums.slimdevices.com/member.php?userid=13777 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 _______________________________________________ audiophiles mailing list [email protected] http://lists.slimdevices.com/mailman/listinfo/audiophiles
