Figure 1.
Stimuli. A, Signal processing block diagram. Signals were low-pass filtered at 6 kHz, sampled at 16 kHz, and quantized with 16-bit resolution. The frequency spectrum of the speech signal was partitioned into 14 frequency bands with a linear-phase finite impulse response filter bank (slopes 60 dB/100 Hz or greater), spanning the range 0.1–6 kHz, spaced in 1/3 octave steps (approximately critical band-wide) across the acoustic spectrum. The Hilbert transform was used to decompose the signal in each band into a slowly varying temporal envelope and a rapidly varying fine structure. The temporal envelope was subsequently low-pass filtered with a cutoff frequency of 40 Hz and then either low- (0–4 Hz) or band- (22–40 Hz) pass filtered. The time delays, relative to the original signal, introduced by the filtering, were compensated by shifting the filter outputs. After the filtering, the envelope was combined with the carrier signal (fine structure) by multiplying the original band by the ratio between the filtered and original envelopes. The result for each original signal (S) is Slow and Shigh, containing only low- or high-modulation frequencies. B, Schematic representation of modulation spectrum for each stimulus category.