I have made a plugin that analyses the selected audio (mono only) using an
A-weighting filter (by Edgar), time weighting FAST (125 ms) and then outputs
the equivalent and maximum level as a label track. The plugin assumes
44.1kHz sampling frequency now, but I guess that can be fixed.
I have made a preliminary verification against professional acoustics
software with minor differences (+/- 0.1 dB). Other than that I let the code
speak for itself, let me know what you think!
;name "Equivalent and maximum dB(A)..."
;action "Calc. A-weighted equivalent level (LAeq) and maximum level with
time weighting FAST (LAFmax)..."
; Mikael Ogren, mr.ogren@...
; Licensed under GPL, no warranty, use at your own risk...
; Calibration so that a 1000 Hz tone with amplitude 1.0 gives 94 dB
(setq calibration (+ 94 28.2))
; A-weighting by Edgar (thanks!)
(setq sa (lp (lp (hp (hp (hp (hp s 20.6) 20.6) 107.7) 737.9) 12200) 12200) )
; Exponential time-weighting filter FAST (125 ms)
; snd-avg is used to downsample to 100 Hz (by averaging over 441 samples)
; This only works for 44.1 kHz sampling frequency, perhaps someone can help
; by making a more general approach that works for all sampl. frq?
; The filtering part is OK for all frequencies, but the "441" constant is
; The constant 0.000001 is to avoid clipping at filtered squared pressure >
(snd-avg (snd-biquad (mult sa sa ) 1 0 0 (exp (/ 1 (mult (snd-srate sa) -
0.125))) 0 0 0) 441 441 OP-AVERAGE)
; Length of the downsampled pressure squared signal
(snd-length saf2 99999999999)
; Calc. the equivalent level
(+ calibration (* 0.5 (linear-to-db (snd-maxsamp (snd-avg saf2 mlength
mlength OP-AVERAGE) ))))
; Calc. the maximum level
(+ calibration (* 0.5 (linear-to-db (snd-maxsamp saf2))))
; Set the output format to 3 digits (example: 53.3 dB)
(setq *float-format* "%#3.3g");
; Output result as a label track (or append into existing label track)
(setq u (format NIL "LAeq= ~A LAFmax= ~A" leq lmax))
(list (list 0.0 u))