fix sampling rate in a-format calibrated soundfiles

calibration/src/sphear_af_calibrate.m

@@ -61,10 +61,12 @@ function [] = sphear_af_calibrate(measurements, keep, fade);
   for m = measurements
     %% read in capsule signal and normalize it
     for c = 1:4
+      printf("processing capsule %d\n", c);
       sig = wavread(["../data/current/af/a-format-" sprintf("%03d", m) "-" char(capnames(c)) ".wav"]);
       cal_sig(:,c) = conv(trunc, sig);
-      wavwrite(cal_sig(:,c), 32, ["../data/current/af/a-format-cal-" sprintf("%03d", m) "-" char(capnames(c)) ".wav"]);
+      wavwrite(cal_sig(:,c), Fs, ["../data/current/af/a-format-cal-" sprintf("%03d", m) "-" char(capnames(c)) ".wav"]);
+      printf("end capsule %d\n", c);
     end
-    wavwrite(cal_sig, 32, ["../data/current/af/a-format-cal-" sprintf("%03d", m) ".wav"]);
+    wavwrite(cal_sig, Fs, ["../data/current/af/a-format-cal-" sprintf("%03d", m) ".wav"]);
   end
 end