I guess there's a math foundation here that would justify the need
for a -1..1 correct signal range...
Is just a convenience.
16bit or 24bit audio files have totally different numbers to represent
the start, middle and end of the same sine wave.
Why would you want Pd to have 0 as one side of the sample's limits?
Mensaje telepatico asistido por maquinas.
On 3/2/2020 12:42 AM, Fede Camara Halac wrote:
Thanks Christof and Lucas for your replies!
The Amp and the Speaker will just work on "one half" of their normal
area. (50% of its audio loudness).
Exactly, that is the difference I hear in loudness.
It shouldn't hurt the amp or the speaker as long as you don't try to
pump up the volume. You might break the speaker if you increase too
much the volume. (it will sound soft but you are pushing it too much
to one of its extremes).
Indeed, this is an important warning that should be in the docs (if
it's there other than getting rid of DC offset, I must have missed it)
Personally, I like to play it safe and add a [hip~ 5] as a DC filter
before my [dac~] :-)
I do this, too, just by (blindly) following the pd manual and
helpfiles... perhaps this calls for a tiny adjustment to the [dac]
object to always implement a hipass (with perhaps a flag to revert to
a non-hipass-5 enabled [dac]). Just a thought, but one that could be
easily argued against given pd's agnostic qualities.
You might use your DAC to send control voltages for a modular
synthesizer, for example.
I have never tried this, but I guess it might be an edge case, and
quite an interesting one.
So, if I want to get more technical as to what exactly happens to the
speakers when sending such "malformed" or "halfformed?" signals, do
you know where I can find good sources that would explain this? I
guess there's a math foundation here that would justify the need for a
-1..1 correct signal range... an age old one that probably relates to
the Nyquist theorem... but I can't see how.
In any case, thanks again for the quick replies!
Best,
f
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