> for linear interpolation, if you are a delayed by 3.5 samples and you
keep that delay constant, the transfer function is
>
>   H(z)  =  (1/2)*(1 + z^-1)*z^-3
>
>that filter goes to -inf dB as omega gets closer to pi.

Note that this holds for symmetric fractional delay filter of any odd order
(i.e., Lagrange interpolation filter, windowed sinc, etc). It's not an
artifact of the simple linear approach, it's a feature of the symmetric,
finite nature of the fractional interpolator. Since there are good reasons
for the symmetry constraint, we are left to trade off oversampling and
filter order/design to get the final passband as flat as we need.

My view is that if you are serious about maintaining fidelity across the
full bandwidth, you need to oversample by at least 2x. That way you can fit
the transition band of your interpolation filter above the signal band. In
applications where you are less concerned about full bandwidth fidelity,
oversampling isn't required. Some argue that 48kHz sample rate is already
effectively oversampled for lots of natural recordings, for example. If
it's already at 96kHz or higher I would not bother oversampling further.

Also this is recommended reading for this thread:

https://ccrma.stanford.edu/~jos/Interpolation/

E

On Tue, Aug 18, 2015 at 1:45 PM, Tom Duffy <tdu...@tascam.com> wrote:

> In order to reconstruct that sinusoid, you'll need a filter with
> an infinitely steep transition band.
> You've demonstrated that SR/2 aliases to 0Hz, i.e. DC.
> That digital stream of samples is not reconstructable.
>
> On 8/18/2015 1:28 PM, Peter S wrote:
>
> That's false. 1, -1, 1, -1, 1, -1 ... is a proper bandlimited signal,
>> and contains no aliasing. That's the maximal allowed frequency without
>> any aliasing. It is a bandlimited Nyquist frequency square wave (which
>> is equivalent to a Nyquist frequency sine wave). From that, you can
>> reconstruct a perfect alias-free sinusoid of frequency SR/2.
>>
>
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