Abstract: The invention relates to a method for characterizing the effects of coupling of a radiofrequency transceiver apparatus comprising at least one transmit path and at least one receive path, and to an apparatus implementing the method. The method comprises the calculation of coefficients of a correcting filter, with the steps of: transmitting a known signal over a transmit path, receiving a signal over a receive path, calculating the coefficients of the correcting filter on the basis of the known signal and of the signal received over said receive path. The method further comprises a step, carried out during the transmission of a useful signal over the transmit path, of filtering the signal transmitted over the transmit path by means of said correcting filter in order to determine the transmitted signal received by coupling effect over the receive path.

Abstract: The invention relates to a method for simplifying a sampled signal digital filter, the method including at least one step for: in order to obtain a first intermediate filter, gathering channels including discrete nonstationary operations relating to the same signal, the first channels including the nonstationary operations relating to a first signal and the second channels including the nonstationary operations relating to a second signal, in order to obtain a second intermediate filter, on each of the first channels and second channels, commutative stationary operations with the nonstationary operations, in order to eliminate the redundant nonstationary operations, and building the filter corresponding to the last obtained intermediate filter.

Abstract: The invention relates to a method for filtering a numerical input signal sampled at a sampling frequency in order to obtain a filtered signal, the method including at least one step for: obtaining a first (respectively second) output signal by carrying out first (respectively second) operations on the first (respectively second) processing channel, the first (respectively second) operations including at least the application of a discrete Fourier transform to M/2p points on a signal coming from the input signal, the integer p being greater than or equal to 1, applying an inverse discrete Fourier transform to M/2p points on the first signal in order to obtain M points of the spectrum of the first signal, M being an integer strictly greater than 2, the application step being carried out by the addition of the results of two processing channels.

Abstract: The device transforms an analog input signal with N components x(0), x(1) . . . x(N?1) into an output signal with N components X(0), X(1) . . . X(N?1) functions of said components x(0), x(1) . . . x(N?1), said device comprising basic addition and/or subtraction cells linked in a butterfly-type architecture to perform said functions, each basic cell comprising an operator performing a conditional division by two of the result of the addition operation or subtraction operation performed, the condition being the saturation of said operator.

Abstract: The invention relates to a method for characterizing the effects of coupling of a radiofrequency transceiver apparatus comprising at least one transmit path and at least one receive path, and to an apparatus implementing the method. The method comprises the calculation of coefficients of a correcting filter, with the steps of: transmitting a known signal over a transmit path, receiving a signal over a receive path, calculating the coefficients of the correcting filter on the basis of the known signal and of the signal received over said receive path. The method further comprises a step, carried out during the transmission of a useful signal over the transmit path, of filtering the signal transmitted over the transmit path by means of said correcting filter in order to determine the transmitted signal received by coupling effect over the receive path.

Abstract: The invention relates to a method for simplifying a sampled signal digital filter, the method including at least one step for: in order to obtain a first intermediate filter, gathering channels including discrete nonstationary operations relating to the same signal, the first channels including the nonstationary operations relating to a first signal and the second channels including the nonstationary operations relating to a second signal, in order to obtain a second intermediate filter, on each of the first channels and second channels, commutative stationary operations with the nonstationary operations, in order to eliminate the redundant nonstationary operations, and building the filter corresponding to the last obtained intermediate filter.

Abstract: The invention relates to a method for filtering a numerical input signal sampled at a sampling frequency in order to obtain a filtered signal, the method including at least one step for: obtaining a first (respectively second) output signal by carrying out first (respectively second) operations on the first (respectively second) processing channel, the first (respectively second) operations including at least the application of a discrete Fourier transform to M/2p points on a signal coming from the input signal, the integer p being greater than or equal to 1, applying an inverse discrete Fourier transform to M/2p points on the first signal in order to obtain M points of the spectrum of the first signal, M being an integer strictly greater than 2, the application step being carried out by the addition of the results of two processing channels.

Abstract: The invention relates to a method for filtering a numerical input signal sampled at a sampling frequency in order to obtain a filtered signal, the method including at least one step for: obtaining a first (respectively second) output signal by carrying out first (respectively second) operations on the first (respectively second) processing channel, the first (respectively second) operations including at least the application of a discrete Fourier transform to M/2 points on a signal coming from the input signal, applying an inverse discrete Fourier transform to M/2 points on the first signal in order to obtain M points of the spectrum of the first signal, M being an integer strictly greater than 2, the application step being carried out by the addition of the results of two processing channels.

Abstract: A wave transmitted by an antenna made up of an array of radiating elements, two pulse waves are transmitted, each modulated by a phase shift law known as modulation phase shift, the phase shifts being in opposition, a first wave being transmitted by a sub-array of radiating elements referred to as odd and the second wave being transmitted by a second sub-array of radiating elements referred to as even, the two sub-arrays being interleaved, the transmitted wave being the combination of the first wave and the second wave.

Abstract: A signal filtering method for filtering a numerical input signal that is sampled at a sampling frequency in order to obtain a filtered signal, the method including a first operation of application of a discrete Fourier transform to M points on a processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the even-numbered indices of a spectral analysis at 2*M points of the processed signal; and a second operation of application of a discrete Fourier transform to M points on the processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the odd-numbered indices of a spectral analysis at 2*M points of the processed signal.

Abstract: A method for determining parameters of a finite impulse response pulse compression filter, implemented by a multi-channel radar comprises: a step Etp10 of transmitting a calibration signal and of acquiring this calibration signal after propagation through the transmission channel, a step Etp20 of injecting the signal acquired, at the input of each of the reception channels, a step Etp30 of measuring the signal at the output of each reception channel, a step Etp40 of calculating the transfer function of the matched filters on the basis of the signals at the output of the reception channels, a step Etp50 of measuring the value of the average power at the output of the various reception channels and of calculating the relative gains between each of the reception channels and a predetermined reception channel on the basis of the measured values of average powers.

Abstract: A wave transmitted by an antenna made up of an array of radiating elements, two pulse waves are transmitted, each modulated by a phase shift law known as modulation phase shift, the phase shifts being in opposition, a first wave being transmitted by a sub-array of radiating elements referred to as odd and the second wave being transmitted by a second sub-array of radiating elements referred to as even, the two sub-arrays being interleaved, the transmitted wave being the combination of the first wave and the second wave.

Abstract: A signal filtering method for filtering a numerical input signal that is sampled at a sampling frequency in order to obtain a filtered signal, the method including a first operation of application of a discrete Fourier transform to M points on a processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the even-numbered indices of a spectral analysis at 2*M points of the processed signal and a second operation of application of a discrete Fourier transform to M points on the processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the odd-numbered indices of a spectral analysis at 2*M points of the processed signal.

Abstract: A method for determining parameters of a finite impulse response pulse compression filter, implemented by a multi-channel radar comprises: a step Etp10 of transmitting a calibration signal and of acquiring this calibration signal after propagation through the transmission channel, a step Etp20 of injecting the signal acquired, at the input of each of the reception channels, a step Etp30 of measuring the signal at the output of each reception channel, a step Etp40 of calculating the transfer function of the matched filters on the basis of the signals at the output of the reception channels, a step Etp50 of measuring the value of the average power at the output of the various reception channels and of calculating the relative gains between each of the reception channels and a predetermined reception channel on the basis of the measured values of average powers.

Abstract: In a method for correcting amplitude and phase offsets in a sigma-delta modulator having a loop including an integrator with a filter and an amplifier, an analog-to-digital converter, a digital-to-analog converter, and an adder-subtractor, an open-loop amplitude/phase frequency response of the modulator is estimated digitally. A phase offset value and an amplitude offset value are calculated by comparing the estimated open-loop amplitude/phase response with a reference amplitude/phase frequency response. The phase and gain offsets of the loop are then compensated for according to the estimated offset values. A sigma-delta modulator implements the method.

Abstract: In a method for improving resolution and for correcting distortions for a sigma-delta modulator, a modulator converts an analog input signal into a secondary output digital signal sampled at a frequency fe and coded on NB bits, a second main output digital signal s?(t) is represented on NMSB bits also being available at the output. At least three processings are applied successively to the outputs, a first processing carrying out a demodulation by a frequency f0 and a decimation of factor N in an independent manner, z second processing carrying out an improvement of the resolution and a third processing carrying out a correction of the distortions. These three processings are carried out after decimation. A sigma-delta modulator implements the method.

Abstract: In a method for controlling the loop delay in a sigma-delta modulator having a loop including an integrator, an analog-to-digital converter, a digital-to-analog converter, and an adder-subtractor, at least one phase-control programmable digital command representing a phase shift is applied to one of the clock signals of the converters of the loop to adjust the relative phase between a clock signal of the analog-to-digital converter and a clock signal of the digital-to-analog converter. A sigma-delta modulator for converting an analog signal implements the method.

Abstract: In a method for correcting amplitude and phase offsets in a sigma-delta modulator having a loop including an integrator with a filter and an amplifier, an analog-to-digital converter, a digital-to-analog converter, and an adder-subtractor, an open-loop amplitude/phase frequency response of the modulator is estimated digitally. A phase offset value and an amplitude offset value are calculated by comparing the estimated open-loop amplitude/phase response with a reference amplitude/phase frequency response. The phase and gain offsets of the loop are then compensated for according to the estimated offset values. A sigma-delta modulator implements the method.

Abstract: The present invention relates to a sigma-delta modulator to convert an analog signal into a digital signal using an analog-to-digital converter slaved in a closed loop. The undecided bits at the output of the analog-to-digital converter are assigned the same values in the digital output signal from the modulator as in the digital signal returned to the input of the modulator.

Abstract: In a method for improving resolution and for correcting distortions for a sigma-delta modulator, a modulator converts an analog input signal into a secondary output digital signal sampled at a frequency fe and coded on NB bits, a second main output digital signal s?(t) is represented on NMSB bits also being available at the output. At least three processings are applied successively to the outputs, a first processing carrying out a demodulation by a frequency f0 and a decimation of factor N in an independent manner, z second processing carrying out an improvement of the resolution and a third processing carrying out a correction of the distortions. These three processings are carried out after decimation. A sigma-delta modulator implements the method.