Abstract: A method for controlling the distortion generated by a system having at least one loudspeaker on board a vehicle, the loudspeaker being designed to receive an audio signal. The method includes measuring at least one indicator of the distortion of the at least one loudspeaker, determining an acceptable distortion threshold for each distortion indicator, which can be used to determine a maximum acceptable amplitude for each frequency in a frequency range of interest of the audio signal entering the loudspeaker system, generating an assembly comprising at least one filter, correcting the audio signal entering the loudspeaker system by applying at least one filter determined in the generating step.

Abstract: A method for controlling the distortion generated by a system having at least one loudspeaker on board a vehicle, the loudspeaker being designed to receive an audio signal. The method includes measuring at least one indicator of the distortion of the at least one loudspeaker, determining an acceptable distortion threshold for each distortion indicator, which can be used to determine a maximum acceptable amplitude for each frequency in a frequency range of interest of the audio signal entering the loudspeaker system, generating an assembly comprising at least one filter, correcting the audio signal entering the loudspeaker system by applying at least one filter determined in the generating step.

Abstract: A signal processor for processing a digital input signal including samples sampled at a sampling frequency, the signal processor comprising a plurality of filters arranged to divide the digital input signal into a first signal in a first frequency band below a first cut-off frequency, and a second signal in a second frequency band above a second cut-off frequency; first frequency shifting circuitry arranged to shift the second signal to a frequency band below the first cut-off frequency; decimation circuitry arranged to decimate the first signal and the shifted second signal; and processing circuitry arranged to process the decimated first and second signals.

Abstract: A telecommunication terminal receives an analog speech signal from a user of the terminal and converts the analog speech signal into a digital signal. A vocoder applies source coding to the speech signal and extracts reconstruction parameters from the speech signal. The reconstruction parameters are modified so that the transmitted voice associated with the signal is modified.

Abstract: A signal processor for processing a digital input signal including samples sampled at a sampling frequency, the signal processor comprising a plurality of filters arranged to divide the digital input signal into a first signal in a first frequency band below a first cut-off frequency, and a second signal in a second frequency band above a second cut-off frequency; first frequency shifting circuitry arranged to shift the second signal to a frequency band below the first cut-off frequency; decimation circuitry arranged to decimate the first signal and the shifted second signal; and processing circuitry arranged to process the decimated first and second signals.

Abstract: A telecommunication terminal receives an analog speech signal from a user of the terminal and converts the analog speech signal into a digital signal. A vocoder applies source coding to the speech signal and extracts reconstruction parameters from the speech signal. The reconstruction parameters are modified so that the transmitted voice associated with the signal is modified.

Abstract: The present invention relates to a telecommunication terminal for generating a customized melody from an analog audio signal recorded by the user, more particularly one suitable for mobile telephony. The terminal comprises input means for receiving an analog audio signal, that includes at least one note, means for sampling and converting said analog audio signal into a digital signal and means for extracting from said digital signal at least three parameters representative of said analog audio signal, these being the frequency, the start time and the duration of said note. According to the invention, it includes means for modifying at least one of the three parameters so as to correct the sound imperfections associated with the quality or with the processing of said analog audio signal.

Abstract: A process for storing audio signals, in particular speech messages, comprises the following process steps: (a) digitalization of incoming audio signals; (b) storage of the digitalized audio signals in a memory in areas with a first memory size and bit rate; (c) monitoring of the occupancy of the memory; (d) determination of the current occupancy rate, in particular full occupancy of the memory; (e) reduction of the memory size and bit rate of the already stored audio signals to a second, smaller value as soon as a predetermined occupancy rate of the memory is reached and (f) occupation of the memory space released in the memory at least in part by newly incoming audio signals.

Abstract: Method and apparatus for detecting voice activity in an audio signal, the method comprising computing the autocorrelation coefficients of the signal, identifying a first autocorrelation vector whose components comprise a first series of autocorrelation coefficients, identifying a second autocorrelation vector whose components comprise a second series of autocorrelation coefficients offset from the first series by a predetermined offset value, subtracting the first autocorrelation vector from the second autocorrelation vector to obtain a differentiation vector, and computing a norm of the differentiation vector, which differentiation vector norm represents a first indicator of voice activity.

Abstract: The invention provides a method of suppressing a background noise signal in a sampled noisy voice signal. The method comprises the following steps: digital frequency-domain processing of the noisy voice signal to produce time-domain filtering coefficients, and digital time-domain processing of the noisy voice signal in accordance with the filter coefficients to produce a voice signal in which the background noise signal is substantially suppressed.