>> well Peter, here again is where you overreach.  assuming, without loss
>> of generality that the sampling period is 1, the continuous-time signals
>>
>>     x(t)  =  1/cos(theta) * cos(pi*t + theta)
>>
>> are all aliases for the signal described above (and incorrectly as
>> "contain[ing] no aliasing").
>
>Well, strictly speaking, that is true. But I assumed the signal to be
>bandlimited to 0..SR/2. In that case, you can perfectly reconstruct
>it, as you have no other alias between 0..SR/2.

That class of signals is band limited to SR/2. The aliasing is in the
amplitude/phase offset, not the frequency.

There are an infinite number of combinations of amplitude/phase of a
nyquist-frequency sinusoid that will all result in the same sampled
sequence. So you can't invert the sampling.

You can construct a DAC that will output some well-behaved nyquist
frequency sinusoid when presented with the input ..., 1, -1, 1, -1, 1, ...,
but you can't guarantee that it will resemble an analog sinusoid that was
sampled to produce such a digital sequence. You don't have enough info to
disambiguate the phase and amplitude.

E

On Tue, Aug 18, 2015 at 1:51 PM, Peter S <peter.schoffhau...@gmail.com>
wrote:

> On 18/08/2015, robert bristow-johnson <r...@audioimagination.com> wrote:
> > On 8/18/15 4:28 PM, Peter S wrote:
> >>
> >> 1, -1, 1, -1, 1, -1 ... is a proper bandlimited signal,
> >> and contains no aliasing. That's the maximal allowed frequency without
> >> any aliasing.
> >
> > well Peter, here again is where you overreach.  assuming, without loss
> > of generality that the sampling period is 1, the continuous-time signals
> >
> >     x(t)  =  1/cos(theta) * cos(pi*t + theta)
> >
> > are all aliases for the signal described above (and incorrectly as
> > "contain[ing] no aliasing").
>
> Well, strictly speaking, that is true. But I assumed the signal to be
> bandlimited to 0..SR/2. In that case, you can perfectly reconstruct
> it, as you have no other alias between 0..SR/2.
>
> -P
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