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