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<blockquote type="cite"> I guess there's a math foundation here
that would justify the need for a -1..1 correct signal range...</blockquote>
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<br>
Is just a convenience.<br>
<br>
16bit or 24bit audio files have totally different numbers to
represent the start, middle and end of the same sine wave.<br>
<br>
Why would you want Pd to have 0 as one side of the sample's
limits?</p>
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<pre class="moz-signature" cols="72">Mensaje telepatico asistido por maquinas.</pre>
<div class="moz-cite-prefix">On 3/2/2020 12:42 AM, Fede Camara Halac
wrote:<br>
</div>
<blockquote type="cite" cite="mid:6662BD22-9D3C-497E-9173-7A1C30364B18@gmail.com">
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<div dir="ltr">Thanks Christof and Lucas for your replies!
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<blockquote type="cite">
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<p>The Amp and the Speaker will just work on "one
half" of their normal area. (50% of its audio
loudness).</p>
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<div>Exactly, that is the difference I hear in
loudness. </div>
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<p>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).</p>
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<div>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) </div>
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<div>
<blockquote type="cite"><span style="-webkit-text-size-adjust: auto;
background-color: rgb(255, 255, 255);">Personally,
I like to play it safe and add a [hip~ 5] as a DC
filter before my [dac~] :-)</span></blockquote>
<br>
</div>
<div><span style="-webkit-text-size-adjust: auto;
background-color: rgb(255, 255, 255);">I do this,
too, just by (blindly) following the pd manual and
helpfiles... </span>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. </div>
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<div>
<blockquote type="cite"><span style="-webkit-text-size-adjust: auto;
background-color: rgb(255, 255, 255);">You might
use your DAC to send control voltages for a
modular synthesizer, for example. </span></blockquote>
<br>
I have never tried this, but I guess it might be an
edge case, and quite an interesting one. </div>
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<div><br>
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<div>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.</div>
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<div><br>
</div>
<div>In any case, thanks again for the quick replies!</div>
<div><br>
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<div>Best,</div>
<div><br>
</div>
<div>f</div>
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