Abstract: A signal processing method transforms X input signals into M output signals. The method includes applying a decorrelation filter to the X input signals so as to generate N decorrelated signals ei; and for each decorrelated signal ei determined, generating N delayed signals ei,k by applying a delay ?k and a gain gk to each decorrelated signal ei, the delays ?k and the gains gk being chosen. For each decorrelated signal ei, the delayed signals ei,k simulate a propagation of the decorrelated signal in a virtual space. The virtual space has N virtual loudspeakers and a virtual listening position that are distributed according to a predetermined geometry. For each value of k, summing the delayed signals ei,k: Ek=?i=1Nei,k; and determining M output signals, each output signal resulting from a linear combination of the N sums of the delayed signals Ek.

Abstract: A method for processing a sound signal including synchronously acquiring an input sound signal Sinput by means of at least two omnidirectional microphones, encoding the input sound signal Sentréeinput in a sound data D format of the ambisonics type of order R, R being a natural number greater than or equal to one, the encoding step including a directivity optimisation sub-step carried out by means of filters of the Finite Impulse Response filter type. Each of the signals acquired by the microphones is filtered during the directivity optimisation sub-step by a FIR filter, then subtracted from an unfiltered version of each of the other signals in order to obtain N enhanced signals. The present invention also relates to a system for processing the sound signal.

Abstract: A method for processing a sound signal including synchronously acquiring an input sound signal Sinput by means of at least two omnidirectional microphones, encoding the input sound signal Sentréeinput in a sound data D format of the ambisonics type of order R, R being a natural number greater than or equal to one, the encoding step including a directivity optimisation sub-step carried out by means of filters of the Finite Impulse Response filter type. Each of the signals acquired by the microphones is filtered during the directivity optimisation sub-step by a FIR filter, then subtracted from an unfiltered version of each of the other signals in order to obtain N enhanced signals. The present invention also relates to a system for processing the sound signal.

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 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 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 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 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.

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 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 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 relates to a method for the sound processing of a stereophonic signal inside a motor vehicle. In a first implementation (“driver” mode) the stereophonic sound source is centered in the middle of the dashboard for the ‘driver’ listen position. For this purpose, delays (t1-t4) are introduced into the frequency bands of the channels transmitted by the speakers, such that the driver appears to be at the center of a circle on which the car speakers are positioned. In a second implementation (“all passengers” mode), the phases of the signals of the two front channels are equalized, such that the sound source appears to be centered on the driver and the front passenger of the vehicle.

Abstract: 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.

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 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 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.