Abstract: A technique for correcting the voice spectral deformations introduced by a communication network. Prior to the operation of equalization of the voice signal of a speaker, the constitution of classes of speakers is communicated, with one voice reference per class. Then, for a given speaker, the classification of this speaker is communicated, that is to say his allocation to a class from predefined classification criteria in order to make a voice reference which is closest to his own correspond to him. Then, for that given speaker, communicating the equalization of the digitized signal of the voice of the speaker carried out with, as a reference spectrum, the voice reference of the class to which the speaker has been allocated. This technique applies to the correction of the timbre of the voice in switched telephone networks, in ISDN networks and in mobile networks.

Abstract: A device for processing digital data. A module (M2, M3) produces on a data vector of the frequency domain Z(k), wherein K varies from 0 to N?1, a convolution with a function U, convolution which corresponds to a cancellation in the time domain of the samples of the inverse transform of Z(k). The function U is in the form: U(k)=sin c(k?k0/2.e)?j?(?(k?k0/2.P(k)), wherein K0 is a constant integer and P(k) a weighting window symmetrical about k0.

Abstract: A technique for correcting the voice spectral deformations introduced by a communication network. Prior to the operation of equalisation of the voice signal of a speaker, the constitution of classes of speakers is communicated, with one voice reference per class. Then, for a given speaker, the classification of this speaker is communicated, that is to say his allocation to a class from predefined classification criteria in order to make a voice reference which is closest to his own correspond to him. Then, for that given speaker, communicating the equalisation of the digitised signal of the voice of the speaker carried out with, as a reference spectrum, the voice reference of the class to which the speaker has been allocated. This technique applies to the correction of the timbre of the voice in switched telephone networks, in ISDN networks and in mobile networks.

Abstract: In order to cancel echo on several sound signal channels, the method consists in generating on each channel a synthetic signal having the spectral characteristics of a white noise; computing the frequency masking thresholds in several adjacent frequency bands using auditory perception properties; bringing in each frequency channel the synthetic signal level to the associated masking threshold value, so as to obtain an auxiliary signal, which is added to the signal associated with the loudspeaker of the channel concerned; supplying the resulting signal to an adaptive filter; adding the output signals of the associated adaptive filters to each microphone channel; subtracting the resulting signal received by the microphone of this channel; deducing an estimation error, on the basis of which the coefficients of the adaptive filters associated with this channel are corrected.

Abstract: A method and a device adapted to hands-free mobile radiotelephony for the optimized processing of a disturbing signal during a sound capture, on the basis of an observation signal y(t) formed of an original useful signal s(t) and of this disturbing signal p(t), the disturbing signal is estimated as a signal p(t) and the useful signal as an estimated useful signal su. An optimal filtering of the observation signal y(t) is carried out on the basis of the signal p(t) and of a minimizing of the error e(su,su) between the useful signal su and the estimated useful signal su. The estimated useful signal su and the useful signal converge towards the original useful signal s(t) for a substantially zero error e(su,su).

Abstract: An echo cancelling device for use between a line receiving an incoming signal and a line transmitting an outgoing signal, for cancelling out echo, comprising a plurality of processing channels connected in parallel relation and assigned to successive mutually adjacent sub-bands of the spectral band of the outgoing signal, each channel having: a first analysis band-pass filter receiving the echo-affected signal to be transmitted, whose output is connected to the additive input of a subtractor; a second analysis band-pass filter, identical to the first filter, receiving the incoming signal and feeding an adaptive filter delivering an estimated echo value in the sub-band to the subtractive input of the subtractor; and a synthesis filter, symmetrical with the analysis filters and whose output feeds the transmission line.

Abstract: An adaptive predicting circuit comprises a lattice filter with N identical cells each having four accesses. The first and second accesses of a cell are respectively connected to the third and fourth accesses of the preceding cell. Each cell comprises a circuit with a delay of one sampling time and two multipliers each having two inputs, one receiving a signal and the other a multiplication coefficient k.sub.m (t). The coefficient k.sub.m (t) is supplied to the multipliers. The adaptive predicting circuit also comprises an adder having N inputs respectively connected to the N cells and one output supplying a prediction signal. The coefficient k.sub.m (t) is supplied by a circuit for the sequential adaptation of the coefficient k.sub.m (t) inherent to each cell. The adaptation circuit has four inputs respectively connected to the first access of the cell, to the output of the delay circuit, and to the third and fourth accesses.

Abstract: A coding process for the adaptive prediction differential PCM type. An error signal e.sub.t is formed between a sample of signal y.sub.t to be coded and a prediction signal p.sub.t of said signal. The prediction signal p.sub.t is formed from restored signals y.sub.t and e.sub.t by two linear filtering operations, the first relating to N successive samples of y.sub.t and using coefficients A1.sub.t, A2.sub.t, . . . , AN.sub.t and the second relating to P successive samples of e.sub.t and using coefficients B1.sub.t, B2.sub.t, . . . , BP.sub.t, the coefficients A1.sub.t, A2.sub.t, . . . , AN.sub.t and B1.sub.t, B2.sub.t, . . . , BP.sub.t being sequentially adjusted at each time t so that the mean power of error signal e.sub.t is minimal. Instead of carrying out the prediction on the basis of signals y.sub.t and e.sub.t only according to the invention, use is also made of the real signal y.sub.t and the real error e.sub.t. A linear filtering of y.sub.t using the coefficients A1.sub.t, A2.sub.t, . . . , AN.sub.

Abstract: Adaptive differential PCM system using both adaptive quantization and adaptive prediction. The adaptive prediction is carried out in adjacent frequency bands on as many signals as are obtained by weighting the quantized signal by coefficients obtained by spectral analysis in the bands of the quantized signal.