Frederic Amadu

The present invention relates to a method for processing the sound signal, comprising the following steps: • Capture synchronously an input sound signal (Sentrée) using microphones N, N being a natural integer greater than or equal to three ,; • encoding said input sound signal (Sentrée) in a data format (D) thereof, said encoding comprising a sub-step of processing said input signal in a serial format type ambisonic R, R being natural integer greater than or equal to one, said converting… Voir plus

The present invention relates to a method for processing the sound signal, comprising the following steps: • Capture synchronously an input sound signal (Sentrée) using microphones N, N being a natural integer greater than or equal to three ,; • encoding said input sound signal (Sentrée) in a data format (D) thereof, said encoding comprising a sub-step of processing said input signal in a serial format type ambisonic R, R being natural integer greater than or equal to one, said converting substep in an ambisonic format type is implemented using a fast Fourier transform, a matrix multiplication, an inverse fast Fourier and using a bandpass filter; and • Return of a sound output signal (Ssortie) using a digital processing of said data (D) of sound. The present invention also relates to a processing system of the sound signal. Voir moins

A method for reducing parasitic vibrations of a loudspeaker environment while maintaining the perception of the low frequencies of an original electric sound signal for broadcast after processing by the loudspeaker having a cut-off frequency. A vibration frequency band is identified that causes the loudspeaker to vibrate. A low frequency band of the original sound signal having a frequency close to the cut-off frequency of the loudspeaker is identified as an upper limit. At least one harmonic… Voir plus

A method for reducing parasitic vibrations of a loudspeaker environment while maintaining the perception of the low frequencies of an original electric sound signal for broadcast after processing by the loudspeaker having a cut-off frequency. A vibration frequency band is identified that causes the loudspeaker to vibrate. A low frequency band of the original sound signal having a frequency close to the cut-off frequency of the loudspeaker is identified as an upper limit. At least one harmonic signal from the isolated low frequency band of the original sound signal is generated. The original sound signal and the harmonic signal are combined to obtain a recombined signal. The vibration frequency band is removed from the recombined signal to obtain a signal for broadcast by the loudspeaker. Voir moins

A method of generating left (Ls) and right (Rs) sound signals known as “surround” signals from a stereo sound signal composed of a left sound signal and of a right sound signal. One of the stereo signals is subtracted from the other signal with the aid of a subtraction module to obtain a single subtraction signal (LR) in which the correlated and in-phase components of the stereo signals (L, R) have been removed. A right sound signal (Rs) and a left sound signal (Ls) are generated from the… Voir plus

A method of generating left (Ls) and right (Rs) sound signals known as “surround” signals from a stereo sound signal composed of a left sound signal and of a right sound signal. One of the stereo signals is subtracted from the other signal with the aid of a subtraction module to obtain a single subtraction signal (LR) in which the correlated and in-phase components of the stereo signals (L, R) have been removed. A right sound signal (Rs) and a left sound signal (Ls) are generated from the subtraction signal (L?R), these signals being decorrelated with respect to one another and corresponding respectively to the right and left sound “surround” signals. These “surround” signals (Ls, Rs) intend for broadcast on the rear channels of an acoustic system. Voir moins

A method for encoding and decoding a digital audio signal composed of an original right-hand signal (SDO) and an original left-hand signal (SGO). The method combines the original right-hand signal (SDO) and the original left-hand signal (SGO) to obtain a single combined signal (SC), encodes the combined signal (SC) using a standard encoder to obtain a compressed combined signal (SCC), and decodes the compressed combined signal (SCC) using a standard decoder (8) to obtain a decompressed combined… Voir plus

A method for encoding and decoding a digital audio signal composed of an original right-hand signal (SDO) and an original left-hand signal (SGO). The method combines the original right-hand signal (SDO) and the original left-hand signal (SGO) to obtain a single combined signal (SC), encodes the combined signal (SC) using a standard encoder to obtain a compressed combined signal (SCC), and decodes the compressed combined signal (SCC) using a standard decoder (8) to obtain a decompressed combined signal (SCD). After decoding, the method generates a reconstructed right-hand signal (SDR) and a reconstructed left-hand signal (SGR) from the decompressed combined signal (SCD), which are de-correlated from each other. Also, a treble-generating module enables the high-frequency component (SHF) of the right-hand (SDR) or left-hand (SGR) signals to be recreated, which signals had been deleted as a result of the compression. Voir moins

A method of optimizing stereo reception for an analog radio by applying the demodulated right sound signal (SD) and left sound signal (SG) as input to a decorrelation module having a variable decorrelation rate. The decorrelation rate of the decorrelation module is modified as a function of the reception quality coefficient “alpha” provided by the radio. The decorrelation module applies a higher decorrelation rate for a smaller reception quality coefficient “alpha” and applies a lower… Voir plus

A method of optimizing stereo reception for an analog radio by applying the demodulated right sound signal (SD) and left sound signal (SG) as input to a decorrelation module having a variable decorrelation rate. The decorrelation rate of the decorrelation module is modified as a function of the reception quality coefficient “alpha” provided by the radio. The decorrelation module applies a higher decorrelation rate for a smaller reception quality coefficient “alpha” and applies a lower decorrelation rate for a larger reception quality coefficient “alpha. Also, a module for generating high-pitched sounds to recreate the high-frequency component (SHF) of the right or left sound signals which has been removed in the event of poor reception. Voir moins

A test platform (11) for facilitating the selection of a sound configuration that is suitable for a target audio system that has a limited processing power (Pmax). During an objective selection of the configurations, the platform (11) adopts—from among a set of possible configurations—the sound configurations that are compatible with the available power (Pmax) of the audio system. Next, the platform (11) makes it possible for an integrator to test the sound rendering of each configuration… Voir plus

A test platform (11) for facilitating the selection of a sound configuration that is suitable for a target audio system that has a limited processing power (Pmax). During an objective selection of the configurations, the platform (11) adopts—from among a set of possible configurations—the sound configurations that are compatible with the available power (Pmax) of the audio system. Next, the platform (11) makes it possible for an integrator to test the sound rendering of each configuration adopted by enabling the selection of the number of virtual loudspeakers and the order (14.2) of the HRTF filters. For this purpose, the integrator can select different types of sound sources to which to listen. After listening to the sound rendering of different sound configurations, the integrator can select the configuration that is most suitable to the target audio system. Voir moins

The invention relates to a method for normalizing the power of an electrical signal, referred to as an original sound signal (S1), said method comprising the steps of: detecting the envelope of the original sound signal (S1); comparing the power value of the envelope signal (S2) with a threshold value (K1); calculating a gain signal (S3) in accordance with said comparison; smoothing the gain signal (S3) so as to obtain a smoothed gain signal (S4); delaying the original sound signal (S1) by a… Voir plus

The invention relates to a method for normalizing the power of an electrical signal, referred to as an original sound signal (S1), said method comprising the steps of: detecting the envelope of the original sound signal (S1); comparing the power value of the envelope signal (S2) with a threshold value (K1); calculating a gain signal (S3) in accordance with said comparison; smoothing the gain signal (S3) so as to obtain a smoothed gain signal (S4); delaying the original sound signal (S1) by a delay (T); and applying the smoothed gain signal (S4) to the delayed original sound signal (S1) such that a normalized sound signal (S5) is obtained. The method of the invention is suitable for a source including a plurality of channels. The invention also relates to a device for implementing the method. Voir moins

L’invention concerne essentiellement un procédé de réduction de vibrations parasites d’un environnement d’un haut-parleur tout en permettant de conserver la perception des basses d’un signal (S1) électrique de son, dit signal sonore orignal, destiné à être diffusé après traitement par ledit haut-parleur (HP) ayant une fréquence de coupure (fc),
caractérisé en ce qu’il comporte les étapes suivantes :
- identifier une bande de fréquences faisant entrer le haut-parleur (HP) en vibrations… Voir plus

L’invention concerne essentiellement un procédé de réduction de vibrations parasites d’un environnement d’un haut-parleur tout en permettant de conserver la perception des basses d’un signal (S1) électrique de son, dit signal sonore orignal, destiné à être diffusé après traitement par ledit haut-parleur (HP) ayant une fréquence de coupure (fc),
caractérisé en ce qu’il comporte les étapes suivantes :
- identifier une bande de fréquences faisant entrer le haut-parleur (HP) en vibrations, dite bande fréquentielle vibratoire,
- isoler une bande (S2) basses fréquences du signal (S1) sonore original ayant comme borne haute une fréquence proche de la fréquence de coupure (fc) du haut-parleur (HP),
- générer au moins un signal d’harmoniques (S3) à partir de la bande basses fréquences (S2) du signal (S3) sonore original isolée,
- combiner le signal (S2) sonore original et le signal (S4) d’harmoniques pour obtenir un signal recombiné (S6),
- supprimer la bande fréquentielle vibratoire du signal recombiné (S6) pour obtenir un signal diffusable par le haut-parleur (HP).
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