Abstract: A method of transmitting convolutionally encoded data with separate, independent and looped encoding over at least one data portion. The data is distributed over one or more cycles, and a plurality of cycles can be grouped into packets for discontinuous transmission if necessary. Weighted decoding is effected independently and cycle by cycle: it starts at a robust location, with a relatively high likelihood, and terminates at a weak location, with a weak likelihood, ignoring the concept of time. This limits the size of the packets of errors and prevents the propagation of packets of errors due to scrambling. Independent encoding and decoding of data can be effected without exchanging parameters between cycles and the parameters of each cycle (size, redundancy, constraint length) can be separate. Different degrees of protection and time-delay are permitted as a function of the nature of the data to be transmitted (voice, digital data, signaling, etc.).

Abstract: A method is disclosed of transmitting convolutionally encoded data with separate, independent and looped encoding over at least one data portion. Thus the data is distributed over one or more cycles. A plurality of cycles can be grouped into packets for discontinuous transmission if necessary. Weighted decoding is effected independently and cycle by cycle: it starts at a robust location, with a relatively high likelihood, and terminates at a weak location, with a weak likelihood, ignoring the concept of time. This limits the size of the packets of errors when errors occur and prevents the propagation of packets of errors due to intentional or non-intentional scrambling. Independent encoding and decoding of data can be effected without exchanging parameters between cycles and the parameters of each cycle (size, redundancy, constraint length) can be separate.

Abstract: A phase-modulated digital signal has N states representing N symbols. The carrier frequency of the signal is estimated through filtering, multiplication of the phase by N, computation of a Fourier transform of the result, and finding the frequency at which the Fourier transform is at a maximum. The frequency can be found by determining the maximum value of the square of the modulus of the Fourier transform and the value of the frequency for which the derivative is zero.

Abstract: The invention concerns a communications system including at least one transmit antenna near at least one receive antenna, transmission producing signals in the receive frequency band. The system includes a transmit neutralizing circuit for supplying a correction signal for eliminating only interfering signals whose frequency is in the receive signal frequency band. The transmit neutralizing circuit is on the input side of an amplifier and comprises means for evaluating, in the wanted receive frequency band, a limited number of parameters of the power amplifier feeding the transmit antenna, this evaluation of parameters being effected on the basis of the input and output signals of the amplifier. The circuit also evaluates the signals causing the interference prior to the input of the amplifier; this evaluation of the interfering signals is based on the calculated parameters and the signal at the input of the amplifier. Application to transmission systems on ships or spacecraft.

Abstract: A method of synthesizing a finite impulse response digital filter includes a phase of calculating the impulse response g(t) of an analog filter the transfer function of which is defined by a predetermined bandwidth B, shape factor SF and out-band attenuation level A. The coefficients of the digital filter are extracted by sampling the resulting impulse response. The calculation phase consists in expanding the impulse response into an integer number M of Hermite functions predetermined in accordance with the shape factor SF. The coefficient extraction phase includes sampling of the expanded impulse response after the calculation phase in a time window the duration of which is dependent on the out-band attenuation level A using a sampling period At that is dependent on the bandwidth B.

Abstract: A method for frame synchronization including the detection of an n-bit pattern with predetermined characteristics in a bitstream is described. The subject method includes the steps of extracting a first set of m bits, with m being smaller than or equal to n, from a first position in said bitstream, deriving from said first set an address of a location in a first memory, deriving from the contents of said location in said first memory at least one second position in said bitstream and at least one second set of bits to be extracted therefrom until said n-bit pattern is detected in said bitstream. An apparatus for performing the subject method is additionally described.

Abstract: A high-throughput filtering method and corresponding apparatus in which each output sample sn is determined by weighting N input samples xi by multiplying the input samples by N coefficients, where the N coefficients are each decomposed into a set of coordinates, each coordinate corresponding to P bits of the coefficient. Essentially, input samples are multiplied by the coordinates of each of the coefficients in a pre-processing operation where the output signals corresponding to the resulting products for a coefficient are added in an adding block. A plurality of adding blocks are provided corresponding respectively to the coefficients such that each block includes the summation result of the products for a corresponding coefficient. The summation result from a previous adding block is added to the summation result of a current block, and the result of this addition is supplied to the next adding block. The output signal is obtained in this manner.

Abstract: An analytical digitizer for signal processing includes a sampler and an analog/digital converter. A first calculating device supplies a first component of the signal made up of a sequence of digital values. Each value is calculated by adding 2h+2 value supplied by the converter weighted with respective non-null integer coefficients a.sub.0, . . . , a.sub.h, a.sub.h, . . . , a.sub.0. A second calculating device supplies a second component of the signal, in phase quadrature to the first, and made up of a sequence of digital values y(t). Each of these values is calculated by adding 2h+2 values supplied by the converter weighted with respective non-null integer coefficients b.sub.0, . . . , b.sub.h-1, -b.sub.h-1, . . . , -b.sub.0. The number and the value of the coefficients are determined by a rigorous method according to the center frequency and the bandwidth of the spectrum of the signal to be processed and according to a fixed limit value for the signal/noise ratio.

Abstract: A method of digitally evaluating the frequency and the phase of signals in the form of digitized samples, the method being wherein it comprises in succession:a stage during which numbers corresponding to the samples of the signals to be analyzed are processed in order to convert them into the form of an analytic signal whose real portion coincides with said signal to be analyzed; and, in parallel therewith:a stage of estimating the parameters to be analyzed which is performed in an overall manner on the basis of estimators and of selection criteria by working on the phase of the signal without using any operator of the Fourrier transform type of any hypothesis test either separately or simultaneously; anda stage of estimating the differences between the real signal as taken in this way and the signal obtained from the estimated parameters, thereby making it possible to deliver data in digital form relating to the quality of the analyzed signal and to the reliability of the estimated values.

Abstract: A method of digitally evaluating the frequency and the phase of signals in the form of digitized samples, the method being wherein it comprises in succession:a stage during which numbers corresponding to the samples of the signal to be analyzed are processed in order to convert them into the form of an analytic signal whose real portion coincides with said signal to be analyzed; and, in parallel therewith:a stage of estimating the parameters to be analyzed which is performed in an overall manner on the basis of estimators and of selection criteria by working on the phase of the signal without using any operator of the Fourier transform type or any hypothesis test either separately or simultaneously; anda stage of estimating the differences between the real signal as taken in this way and the signal obtained from the estimated parameters, thereby making it possible to deliver data in digital form relating to the quality of the analyzed signal and to the reliability of the estimated values.

Abstract: A signal may be demodulated without using any analog components by using a digital processor. A digitizer delivers the amplitude of the signal to be demodulated at a sampling rate. The signal is acquired from the digitizer and stored in a common memory. The amplitude samples are converted to phase samples. Symbol rate phase hypotheses are generated for splitting the time signal into slices. The carrier phase for one symbol is then calculated for each symbol rate phase hypothesis so that the carrier phase can be estimated over a plurality of (S) symbols. The symbol phase is compared to the carrier phase so that the demodulated signal may be extracted and so that the quality of the demodulation may be estimated.

Abstract: The demodulator combines a conventional frequency discriminator with a bandpass filter for removing the noise which is present from the band occupied at a given instant by the modulated signal. The center frequency of the bandpass filter is controlled in dependence on the instantaneous frequency of the modulated signal and its bandwidth b.sub.u is controlled in dependence on the instantaneous width of the band occupied by the modulated signal.

Abstract: A process for the demodulation of an amplitude modulated signal composed of two suppressed carriers, at a same frequency F and in phase quadrature, which are respectively modulated by two modulation signals, the modulated signal being preceded by a reference signal at frequency F, bysampling the reference signal and the modulated signal, according to a frequency m.F (m being an integer);determining a polar angle .psi..sub.t of a rotating vector representing the reference signal, at time t, by calculating: ##EQU1## where x(t) and x(t-(1/4F)) are the values of two successive samples of the reference signal;determining polar coordinates (.rho..sub.t, .theta..sub.t) of a rotating vector representing the modulated signal, at time t, by calculating: ##EQU2## where y(t) and y(t-(1/4F)) are values of two successive samples of the modulated signal,calculating .gamma.=.theta..sub.t -.omega..sub.t -.phi..sub.o, where .phi..sub.o is the phase displacement between the reference signal and one of the carriers,calculating .

Abstract: In a digital communication system in which the information is transmitted by successions of bits termed packets, stations may be called upon to perform the function of a relay between other stations. The stations which may be called upon to perform the function of a relay comprise: a checking device which merely checks the address code of a packet received; an eliminating and switching device which either destroys the packet (doubtful address code) or orients the packet toward receiving means pertaining to the station (address code identical to the code of the station) or toward the station corresponding to the address code by passing through a transit memory; and information producing means for producing packets within the station. The contents of the transit memory are transmitted in priority by the station.