Abstract: A multiplex transmission system is disclosed in which speech activity is detected as a sequence of talkspurts and intervening silent intervals. The speech information is encoded and accumulated in a buffer store by omitting the silent intervals. A time stamp is associated with each talkspurt code burst to permit approximate reconstruction of the talkspurt time structure at the receiver. The talkspurt code bursts are assembled into packets of optimum size for transmission on shared transmission facilities, such as a Time Assignment Speech Interpolation (TASI) system.The assembled talkspurt packets can be encoded by an adaptive technique responsive to the loading on the transmission facilities and can be transmitted at a rate faster than real-time speech generation to accommodate high-level usage on the shared transmission facilities.

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.

Abstract: A loudspeaking teleconferencing circuit serving a plurality of remote conference room locations includes a plurality of speakerphone sets at each conference room location and a communication network connected to an incoming and an outgoing terminal at each location. The transmit channel of each set couples a microphone to the outgoing terminal; the receive channel of each set couples the incoming terminal to a loudspeaking device; and a control circuit selectively activates one of the set transmit and receive channels at a time responsive to speech energy present on the transmit and receive channels. Each set control circuit is operative independently of the control circuits of the other sets; and the outputs of all active transmit channels are combined and applied through the location outgoing terminal to the communication network.

Abstract: Disclosed is a system for synthesizing emphasis-controlled speech from stored signals representative of words precoded by a phase vocoder having analysis bands which are wide relative to the voice harmonic frequency spacings. The stored signals comprise short-time Fourier transform parameters which describe the magnitude and the phase derivative of the short-time speech spectrum. Speech emphasis-controlled synthesis is achieved by extracting the stored signals of chosen words under control of a pitch-duration signal, by concatenating the extracted signals, by modifying the magnitude parameters of the extracted signals to effect a desired speech intensity, by interpolating the extracted parameters, and by decoding the resultant signals in accordance with phase vocoder techniques.

Abstract: Disclosed is a system for synthesizing speech from stored signals representative of words precoded in accordance with phase vocoder techniques. The stored signals comprise short-time Fourier transform parameters which describe the magnitude and phase derivative of the short-time signal spectrum. Speech synthesis is achieved by extracting the stored signals of chosen words under control of a duration factor signal, by concatenating the extracted signals, by operating on the phase derivative parameters to effect a desired speech pitch change, by interpolating the magnitude parameters of the short-time Fourier transform in response to the pitch and duration changes, and by decoding the resultant signals in accordance with phase vocoder techniques.