Abstract: A speech signal amplitude equalizer (compressor) circuit normalizes the amplitude variations of a real-time input speech signal s(t) to form the normalized output signal r(t) which may be either of two values: a constant amplitude (unilevel) speech signal or a zero-amplitude (squelched) signal. The normalized output signal r(t)=s(t)/a(t)+K where the envelope signal a(t)=s.sup.2 (t)+s.sup.2 (t) in which s.sup.2 (t) is the Hilbert transform and K is a control signal which can take one of two levels depending upon the amplitude of the envelope signal being above or below certain thresholds in a hysteresis type logic circuit.

Abstract: A digital speech communication arrangement includes apparatus for partitioning a speech signal into a plurality of nonuniformly related, preassigned intelligibility contributing subbands, and quantized with an accuracy (bit allocation) based upon perceptual criteria. As a result, the quality of the coded signal is improved over that obtained from a single full-band coding of the total spectrum. "Integer-band" sampling may be employed to alias the signal in an advantageous way before coding. Each subband portion is digitally encoded after sampling at a rate related to the bandwidth of the subband. The digital codes of the subband are combined and applied to a common communication channel. A replica of each subband is formed by separating the digital codes, producing a sampled form of each digital coded signal and selecting the subband of the speech signal from said sampled form. The subbands are combined to reconstruct the speech signal. Other features include: aliasing for efficient frequency transposition.