Extract amplitude envelope of filtered speech signal. Adapted from Tilson & Johnson (2008). Procedure:

extract_env(
  x,
  fs,
  low_pass = 80,
  fs_out = 80,
  win = c(700, 1300),
  mean_centre = FALSE,
  replace_init = FALSE
)

Arguments

x

a speech signal

fs

sampling frequency of signal

low_pass

frequency of lowpass filter used for smoothing

fs_out

output sampling frequency

win

lower and upper frequencies for initial bypass filter. Default is 700Hz-1300Hz as in Tilson & Johnson (2008)

mean_centre

if TRUE signal will be scaled between 0 and 1 and then mean centred. Default is FALSE

replace_init

if TRUE (default is FALSE) first sample of result will be replaced with second sample to deal with initialisation issue in resampling

Value

A matrix with time and amplitude

Details

1. Signal is bypass filtered to extract desired frequency range 2. Absolute signal is then lowpass filtered 3. Signal is downsampled and mean centred if desired

References

Tilsen, S., & Johnson, K. (2008). Low-frequency Fourier analysis of speech rhythm. The Journal of the Acoustical Society of America, 124(2), EL34–EL39. doi:10.1121/1.2947626

See also

fft_spectro