Abstract: A synthesis method in which that part of each interval of the original signal which contains the fundamental information is left unchanged, and only the remaining part of the interval is altered. In this way, not only is processing time reduced, but the natural sound of the synthetic signal is also improved. The main part of the interval is an exact reproduction of the original signal. At least the waveforms associated to voiced sounds are subdivided into a plurality of intervals, corresponding to the responses of the vocal duct to a series of excitation impulses of the vocal cords, synchronous with the fundamental frequency of the signal. Each interval is subjected to a weighting. The signals resulting from the weighting are replaced with a replica thereof shifted in time by an amount that depends on a prosodic information. The synthesis is then carried out by overlapping and adding the shifted signals.

Abstract: The method provides the use, for the determination of the estimation echo-channel pulse response, of digital filtering techniques of the transversal adaptive type, and the subdivision of the pulse-response of the equivalent echo channel into three intervals, the second of which consists of the significant portion of the pulse response and is determined by the iterative search for the value and the position of the coefficient of maximum absolute value; the first interval corresponds to a pure delay and the third corresponds to the contribution given by the lowest frequencies of the base band. To copy the echo-signal, there are only considered the pulse-response coefficients whose indexes are comprised in the second interval, and the samples of the signals to be copied belonging to preceding sampling instants, whose indexes are also comprised in the second interval.

Abstract: A multichannel digital speech synthesizer comprises a pulse generator storing periodic and aperiodic excitation signals to be processed in a lattice filter according to weighting parameters, such as gain and reflection coefficients, transmitted from a computer via a control unit and a plurality of input modules assigned to respective output channels. Each input module includes a resettable counter for timing the emissions of periodic or aperiodic excitation signals, to generate a voiced or an unvoiced speech element, and for requesting a new set of parameters from the computer upon detecting the end of a validity interval for a current set of parameters; the module further comprises a pair of buffer memories alternating in reading and writing operations under the control of the counter to ensure a continuous flow of parameter sets to the filter.

Abstract: A transmitting terminal, communicating via a PCM link with a remote receiving terminal, processes two simultaneously arriving bit streams consisting of recurrent frames of 32 time slots, 29 of these time slots containing 8-bit code words which represent voice samples from as many PCM channels. The two arriving bit streams are transcoded by an ADPCM (adaptive differential pulse-code modulation) technique to convert the 8-bit words into substantially equivalent 4-bit words which are then combined, in interleaved relationship, into a single bit stream sent to the receiving terminal, each frame of this composite bit stream having 29 of its 32 time slots occupied by two four-bit words respectively taken from the original bit streams. At the receiving terminal a complementary procedure is followed to separate the two groups of 4-bit code words from each other and to reconvert each of them into an 8-bit word, with substantial reconstitution of the original bit streams.