Cselt-centro Studi E Laboratori Telecomunicazioni S.p.a.

Maurizio Copperi

Daniele Sereno

Luciano Nebbia

Roberto Billi

This method provides a filtering of blocks of digital samples of speech signal by a linear-prediction inverse filter followed by a shaping filter, whose coefficients are chosen out of a codebook of quantized filter coefficient vectors, obtaining a residual signal subdivided into vectors. Each vector is classified by an index q depending on the zero-crossing frequency and r.m.s. value; it is then normalized on the basis of the quantized r.m.s. value, and then of a vector of quantized short-term mean values; the mean-square error made in quantizing said vectors with vectors contained in a codebook and forming excitation waveforms in computed. In this codebook the search is limited to a subset of vectors determined by index q and p of short-term mean vector. The coding signal consists of the index of the filter coefficient vector, of indices q, p, of quantization index m of the r.m.s. value, and of the index of the vector of the excitation waveform which has generated minimum weighted mean-square error (FIG. 1).

Method of and device for speech signal coding and decoding by parameter

According to the coding method, the digital speech signal is split into subbands; each subband is coded independently of the others. The first two subbands undergo an adaptive linear prediction operation yielding a residual, normalized with respect to a quantized r.m.s. value; the residual samples are quantized and coded by a variable bit number. As to the third subband the samples normalized with respect to a quantized gain term are directly quantized and coded by a variable bit number. While splitting the subband signals into sample blocks, for the first two subbands the coefficient vectors of subband linear prediction filters relevant to each block are also determined by vector quantization and linear prediction inverse filtering operations; quantized r.m.s. values are also determined to calculate the numbers of bits permitting subband signals coding. The coded signal consists of subband signal quantization level indices, of r.m.s. value quantization level indices, and of linear prediction filter coefficient vector indices.

Method of and device for speech signal coding and decoding by subband

This method provides a filtering of digital samples of speech signal by a linear-prediction inverse filter, whose coefficients are chosen out of a codebook of quantized filter coefficient vectors, obtaining a residual signal subdivided into vectors. The weighted mean-square error made in quantizing said vectors with quantized residual vectors contained in a codebook and forming excitation waveforms is computed. The coding signal for each block of samples consists of the coefficient vector index chosen for the inverse filter as well as of the indices of the vectors of the excitation waveforms which have generated minimum weighted mean-square error. During the decoding phase, a synthesis filter, having the same coefficients as chosen for the inverse filter, is excited by quantized-residual vectors chosen during the coding phase (FIGS. 1, 2).

Method of and device for speech signal coding and decoding by vector

A four-wire trunk line of a telephone network, operating with frequency-division multiplexing, is coupled by a hybrid coil to a local two-wire line and is provided, at a location separated from that hybrid coil by frequency modulators and demodulators, with an echo canceler connected across its receiving and transmitting branches. In order to eliminate frequency offset resulting from the incoherent modulation and demodulation, the echo canceler coacts with the phase-roll compensator comprising a phase-locking loop driven, during an acquisition phase, from a digital pilot-tone generator temporarily connected via a digital/analog converter to the receiving branch. The loop includes an analog/digital converter in the transmitting branch receiving the echo of the pilot tone due to hybrid-coil unbalance, a 90.degree. phase shifter and a delay line which are connected in parallel to the output of that converter and work into respective multipliers whose output signals are summed in a digital adder, another multiplier heterodyning the adder output with the pilot tone to produce a digital beat frequency, and a function generator feeding sine and cosine functions of that beat frequency to the multipliers that are respectively coupled to the phase shifter and to the delay line. The output signals from the loop adder pass through the echo canceler, which operates on them in a working phase, to another digital/analog converter inserted in the transmitting branch. In another embodiment the tasks of the phase-roll compensator are performed by a microprogrammed processor.

Digital echo-cancellation method and circuit arrangement

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.

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Venaria, Italy

Maurizio Copperi