Abstract: An audio-signal equalisation device for a vehicle using a data communication bus includes a memory storing speed filter data sets comprising triplets associating a speed value and a gain value associated with coefficients for defining a speed masking filter, and functional filter data sets comprising triplets associating an engine or ventilation value and a gain value associated with coefficients for defining an engine or ventilation masking filter. The device further includes a collector receiving parameters accessible on the communication bus and designating the vehicle speed and a functional parameter selected from the vehicle engine speed and a ventilation data set. The device further includes a computer receiving as an input a noise margin value, and the parameters, to derive therefrom a speed filter data set and a functional filter data set and to combine them in a masking equalisation filter to be applied to the audio signal.

Abstract: A method for automated setting of digital processing parameters intended to be applied to digital signals before broadcast by at least one loudspeaker placed in an environment. The method includes a step of determining a set of frequency responses of the environment, an equalization step applied for each subset of at least one loudspeaker and a second equalization step applied to at least one subset of at least one loudspeaker. A device for the implementation of the method is also provided.

Abstract: A system and method of processing a sound signal. The method of processing a sound signal that includes the following steps: Synchronous reception of an input sound signal (Sinput) of N microphones, N being a natural number greater than or equal to three; Encoding of the input sound signal (Sinput) in a data format (D) of sound, the encoding including a sub-step of transforming the input signal into an ambisonic format of order R, R being a natural number greater than or equal to one, the sub-step of transformation into an ambisonic format is carried out by a Fast Fourier Transform, a matrix multiplication, an Inverse Fast Fourier Transform and by a band pass filter; and Return of an output sound signal (Soutput) by digital processing of the sound data (D).

Abstract: A system for optimizing music listening which includes a multimedia server controlled by a user. The multimedia server includes a transmitter of the multimedia server to transmit an audio stream to various connected appliances provided with an acoustic emitter and a selector to select at least one connected appliance to which the audio stream is transmitted. A predetermined table of characteristics links the various connected appliances to a measurement of the acoustic reproduction capabilities thereof. A predetermined user table links at least one user with the hearing capabilities thereof. An adapter to adapt the audio stream in accordance with the measurement of the acoustic reproduction capabilities of the connected appliance to which the audio stream is transmitted and in accordance with the hearing capabilities of the user connected to the multimedia server.

Abstract: A method for suppressing the late reverberation of an audio signal. A plurality of prediction vectors are calculated. A plurality of observation vectors from the modulus of the complex time-frequency transform of an input signal is generated. A plurality of synthesis dictionaries from the plurality of observation vectors are constructed. A late reverberation spectrum from the plurality of synthesis dictionaries and the plurality of prediction vectors are estimated. A plurality of observation vectors are filtered to eliminate the late reverberation spectrum and obtain a dereverberated signal modulus.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method and device for normalizing the power of an electrical signal, referred to as an original sound signal S1. The method detects the envelope of the original sound signal S1 and compares the power value of the envelope signal S2 with a threshold value K1. The gain signal S3 is calculated in accordance with the comparison and smoothed to obtain a smoothed gain signal S4. The original sound signal S1 is delayed by a delay T. The smoothed gain signal S4 is applied to the delayed original sound signal S1 to obtain a normalized sound signal S5. The method is suitable for a source including a plurality of channels.

Abstract: A method and device for normalizing the power of an electrical signal, referred to as an original sound signal S1. The method detects the envelope of the original sound signal S1 and compares the power value of the envelope signal S2 with a threshold value K1. The gain signal S3 is calculated in accordance with the comparison and smoothed to obtain a smoothed gain signal S4. The original sound signal S1 is delayed by a delay T. The smoothed gain signal S4 is applied to the delayed original sound signal S1 to obtain a normalized sound signal S5. The method is suitable for a source including a plurality of channels.

Abstract: A method for selecting a perceptually optimal HRTF in a database according to morphological parameters. A first database includes the HRTFs of subjects M, a second database includes the morphological parameters of the subjects, and a third database corresponds to a perceptual classification of the HRTFs. The N most relevant morphological parameters are sorted by correlating the second and third databases. A multidimensional space is created, which optimizes the spatial separation between the HRTFs according to the classification thereof in the third database to obtain an optimized space. An optimized projection model MPO is calculated for correlating K optimal morphological parameters with the corresponding position of the HRTF filters in the optimized space. For any user whose HRTF is not included in the database, at least one HRTF can be selected from the database BD1 according to the parameters K of the user and the optimized projection model MPO.

Abstract: 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.

Abstract: The invention essentially relates to a method of producing more than two different electric time sound signals (C(t), GF(t), DF(t), GA(t), DA(t)) from two initial electric time signals (GI(t), DI(t)). The inventive method comprises the following steps consisting in: in the frequency domain, producing a central electric frequency sound signal (C(v)) from the in-phase frequency components of the initial signals; and producing two front signals (GF(t), DF(t)) by subtracting the central signal (C(t)) from the initial signals (GI(t), DI(t)) In addition, two rear signals (GA(t), DA(t)) can be produced from the out-of-phase frequency components of the initial signals. In this way, the method can be used to transform a stereophonic signal into a type 5.1 signal comprising five different sound signals.

Abstract: Method intended for the picking up of sounds coming from at least one sound source, for the recording of said sounds and their play-back, by means of a sound pick-up assembly, a recording medium and a transmission assembly. the sound pick-up assembly being composed of at least two microphones, the relative position of which is constant, the recording medium being of any known stereophonic type, particularly with at least two tracks, the transmission assembly comprising at least two members such as earphones, loudspeakers or speaker boxes. It is characterised in that sounds are picked up simultaneously by the two microphones, known as “right” (11) and “left” (12) respectively and that these two microphones (11 and 12) are displaced together with respect to said sound source (A), particularly by causing the distance and/or height of each microphone (11-12) to vary differentially with respect to that source (A), i.e.