Hi guys,
how can I get sound level meter values from [adc~]?
I tried [env~] and [sigmund~] but they give me numbers that are very different from the ones I get from a real SLM.
What kind of scale a SLM does use and how can I obtain similar results with pd?
thanks a lot

Like a SLM, env~ gives numbers in decibels, or dB. dB is a logarithmic scale for signal energy, with a change of 20 dB being equal to a factor of 10 in amplitude.

A dB scale needs a reference to call 0 dB. When the root-mean-squared average of its input is 1, env~ arbitrarily calls that 100 dB, so if you connect an osc~ directly to env~, you will see 97 dB, because the RMS level of a sinusoid is 3 dB down from the peak amplitude. Your SLM should use an RMS level of 20 microPascals as 0 dB, so a 94 dB sound has an RMS of 1 Pascal.

The signal that gets to env~ through your mic, preamp, soundcard, and adc~ is subject to all sorts of scaling that env~ can't know anything about, so you need to figure out a calibration factor.

Measure the same sound with your SLM and with env~. Then figure out what value you need to add or subtract from the env~ value to make it equal to the SLM. This should work for sounds that are similar, but note that your SLM probably applies a frequency weighting (A or C) to the sound before computing the RMS, so if you try to measure sound with very different frequency content, your env~ and SLM measurements will still have discrepancies.

Thank you very much for your clear answer emacpher.
I'm doing that way already (SLM, env~, calibration) but even if the two start at the same value when the dynamic increases the SLM reaches higther values than env~. This must be for the frequency weighting you said. I'll inquire further about that... maybe a filter before the RMS?
thanks

Many factors could cause these differences: mic frequency responses, polar patterns, and noise floors; integration times for RMS averages, and whether the SLM is reporting average or peak levels; possible compression or distortion in your signal chain.

You could avoid all the mic directionality, frequency response, and integration time issues by trying to get things working with a constant pure tone first. If things are not behaving right with that, look at your adc~ signal as you change the tone intensity to see what might be going on.