Other than short.

Yes, but as far as I know, that's a different issue.

It's an interesting question.  I think about it in terms of the magnitude
of the error.  The size of the error will always be smaller than the
magnitude of the least significant bit in the digital sample; otherwise,
you'd be choosing a different digital value.  So you could think of the
result of playing back the series of digital samples as being equivalent
to playing back the original analog values (before quantization) plus
very small values that are the difference of the analog and the digital.

When you think of it in these terms, it's like you're adding a small
noise signal to the original.  But the noise signal's magnitude is so
small that it's barely noticeable (if your samples are high-enough
resolution).  In fact, it may easily be dwarfed by some noise source
that was original analog signal anyway.  In the digital domain, there
is obviously no such thing as a noise-free reproduction of a signal,
because that would require an infinite amount of information.  But the
thing not to miss is that there is no such thing as a noise-free
reproduction of a signal in the analog domain either.  Even a simple
pair of wires adds distortion:  it has capacitance and so acts as a
low-pass filter.  I'm not a professional audio guy or anything, but
from what I understand, in the audio world, any A-to-D with a sample
size larger than 24 bits is viewed as partly a joke, because many people
are convinced nobody sells any equipment with analog circuits that are
quiet enough so that the last few bits are anything other than a
digital representation of analog noise, even at 24 bits.  (Of course,
larger sample sizes are useful when processing the audio.)

Yes, it might not have been all that clear, but I was referring to
the format of the samples in the input file.  They probably are
two's-complement, but in theory they could be something else.  They
could have their bits completely reversed or they could be a gray
code or one's-complement or something.  Unlikely, but you never know.
There are some strange file formats out there.

That's an interesting idea.  I haven't had much occasion to use nio,
and I didn't know it had that stuff in it.

  - Logan

Yes, I believe that's what actually happens in practice.  You effectively
sample at one rate and then downconvert back to rate that you can use to
work with the data and transmit the data.  I'm fairly sure that in most
commercially-available audio hardware, the oversampling and downconversion
happens in the hardware, so that the software and the end-user don't need
to be aware of it.

Actually, that's not even true now that I think of it.  A lot of A-to-D
converters (maybe even the vast majority now?) are "one-bit", which means
they use delta-sigma coding, which involves some analog circuitry to
modulate the signal as they take one-bit samples at a very high rate.
That then gets converted into PCM data at a lower rate.

  - Logan

Aha, well if you're using a software tone generator (which I had not
considered), you may actually be able to see the maximum theoretical
frequency.  But don't expect it to come across as a sine wave.  Since
2000 is an exact multiple of 1000, you will be hitting the sine wave
at the same exact two points in its phase every time, and so the sample
data you see will just be a series of two alternating sample values.

That makes me think a little further:  I would have expected a sequence
of bytes that repeated every two samples.  But the output data you
posted showed variation even after the first four bytes.  This makes
me wonder if you are not loading a file format that has a header in
front of the sample data.  If you think that might be happening, you
could generate about 10 or more seconds worth of data, then skip over
the first few kilobytes of the input file to get past the header.

  - Logan