[LAD] simulating analog audio devices

[Erik de Castro Lopo]

Robin Gareus wrote:

> hehe, right. ATM i don't interpolate at all.
> The input just stays for the duration of the sample.

Thats called zero order hold:

    http://en.wikipedia.org/wiki/Zero_Order_Hold

and is a really crappy sound sample rate converter. Signal to noise
ratio is about 10dB.

> Can I feed the rabbit at irregular intervals: ie. specify time and sample?

No, not really.

When Spice loads the libsndfile component, can the component find out
the inter-sample period used by Spice? If so, the best solution would
be to use the Rabbit to up sample to that inter-sample period.

If the above is not possible, I suggest that you up sample the audio 
by say 128 times and then linearly interpolate to get whatever inter
sample values you need.

> So far I've only found the command to set the *maximum* time-step, not
> an option to enforce it. - setting it to 1/Fsample is ok. setting it to
>   0.25/Fsample and using only about every 4th value gives much better
> results..  using 0.01/Fsample makes me fetch a few more coffees;

I suspect that using 0.01/Fsample will give you a result much closer
to the real device. I suspect that what you get at 1/Fsample produces
results that may not match reality.

Erik
-- 
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Erik de Castro Lopo
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