JohnSwenson wrote: 
> First let me say that I don't have a fully defined set of rules such as
> "if xyz is greater than abc it sounds bad" . Also it's difficult to find
> out exact implementation details of particular filters without the
> designers telling you what they did in the chip. (some do, most don't)
> 
> What I have done is looked at the graphs in the spec sheets and listened
> to many different chips and come to some conclusions, then came up with
> a number of hypothesis and run a bunch of tests to see how these pan
> out, and have come up with some theories based on that. Note these are
> just observations, I have no idea WHY it is this way. 
> 
> Number one theory is the shape of  the passband. If it looks
> "complicated", different frequency bands looking very different, or
> looking like two different simple functions superimposed on each other,
> then the sonics are compromised. All such filters are combinations of
> more than one type of filter. This seems to be the number one issue,
> every test I have done shows that a single simple filter sounds better.
> Exactly WHAT filter is not nearly as important as it being a simple
> filter. 
> 
> The problem is that in order to get very high stop band attenuation in
> reasonable amount of silicon takes such complex filters. And for some
> reason the stop band attenuation has become THE metric companies are
> trying to optimize in their spec sheets. The result is that all embedded
> digital filters are complex types, and hence are sonically compromised.
> My opinion is that this is why there is so much talk about "all the
> chips sound alike", they are all being similarly compromised by these
> complex filters. 
> 
> One interesting aspect of this is that many chips have a complex filter
> for 44.1 and 48, a less complex filter for 88.2 and 96 and a basic
> simple filter or no filter at all for 176.4 and 192. I think this is why
> a lot of people go for upsampling. Unfortunately if you do it in
> hardware, the upsampling is done with a complex filter, so you haven't
> gained much, BUT by doing it in software you can use a high quality
> simple filter. This difference in filter complexness is probably part of
> why there is such interest in the higher sampling rates. 
> 
> Theory two, ultimate stop band attenuation is not all that critical.
> With no filter you can definitely hear the aliasing, I like to
> characterize this as the sound is "dirty". BUT you don't have the issues
> of the complex digital filters. I think this is why NOS (No Over
> Sampling) has a significant following. To get good sound you DO need to
> get rid of the aliasing, but how much? I have found that 60db of stop
> band attenuation is perfectly acceptible to clean up the sound, and you
> can easily achieve that with a simple filter. I have yet to hear a
> difference with any simple filter just by changing the stop band from
> 60db to 120db. But huge differences when switching to a complex filter
> to get that 120db. 
> 
> That's it for the theories. I have a bunch more observations but haven't
> had enough testing to get into theories. I have done some listening to
> things like zero phase, minimum phase, pre ringing, post ringing, long
> time, short time etc. They do make differences, but I have not come to
> any conclusions about what sounds best. The differences these things
> make are small in comparison to the difference between a complex and a
> simple filter. 
> 
> There has been a lot of talk about the  "steepness" of the filter, my
> experience is that in and of iteslf this is not very important, the big
> issue is that again in order to get a very steep filter  most
> implementations use a complex filter. The "gentle slope" filters are
> usually much simpler and THAT it the real difference. 
> 
> Again no hard data on this, it's just me listening to a whole bunch of
> stuff and trying to come up with some correlations. Also no clue WHY
> human hearing seems to be so sensitive to complex filters. 
> 
> John S.
Thanks John
I'm intrigued to try this out, but am not sure how to be sure that i am
getting a simple rather than complex filter. When upsampling in sox what
filter would you recommend - would I be safe specifying a Ilinear phase
filter cutting in at say 18 kHz?


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