<div dir="ltr"><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons!</div></div><div class="gmail_extra">
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<br><br><div class="gmail_quote">On 21 December 2013 13:48, IOhannes m zmölnig <span dir="ltr"><<a href="mailto:zmoelnig@iem.at" target="_blank">zmoelnig@iem.at</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On 2013-12-20 23:34, Martin Peach wrote:<br>
> On 2013-12-20 16:55, Alexandre Torres Porres wrote:<br>
>> Hi there, where can I find info about headroom and clipping on Pd. Or<br>
>> can anyone tell me quickly how it goes?<br>
>><br>
>> Does it always really clip over a maximum of 1, or is there some<br>
>> headroom? Does it depend on the audiocard or something?<br>
><br>
> The soundcard will always clip above +1 and below -1, and sometimes even<br>
<br>
</div>yes, but the output of Pd (what you send to [dac~]) is not necessarily<br>
sent directly to the soundcard: e.g when using jack, the samples will be<br>
passed as floating point values to the next client: so samples exceeding<br>
-1..+1 need not clip at all.<br>
you can confirm this by connecting the output of Pd to the input of Pd<br>
via jack, and send a [osc~ 440]*10 ...<br>
<br>
but when you connect this output to the system output, you will get<br>
clipping.<br>
<br>
afaik, you get something similar on OSX, where the (portaudio) API will<br>
take floating point samples, and the signal gets sent through a limiter<br>
to prevent obvious clipping.<br>
<br>
so the bottom line is: hardware always has a physical range limit (that<br>
is mapped to -1..+1).<br>
but Pd is software and some audio APIs can handle sample values in<br>
floating point format just fine. with these, you most likely don't get<br>
any *immediate* problems if the range exceeds -1..+1.<br>
however, how excessive samples are handled is highly depending on the<br>
audio API and eventually other components in the signal chain.<br>
so if you do want to output signals that are outside -1..+1, you are on<br>
your own (from Pd's perspective).<br>
<div class="im"><br>
<br>
> within those limits (if the interpolated waveform between samples goes<br>
> over the limit).<br>
<br>
</div>well, the reconstruction filters in the DAC won't necessarily clip in<br>
this case.<br>
<br>
<br>
gfamdsr<br>
<span class="HOEnZb"><font color="#888888">IOhannes<br>
<br>
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