Abstract: The headset includes an electro-acoustic transducer, at least one noise reduction processing chain including external and internal microphones, a processing filter including mounted in parallel between the external microphone and an adder, at least one elementary filter each including a recursive filter and an elementary variable gain amplifier, a control unit for the gain of each elementary amplifier according to a signal coming from the external microphone and an error signal coming from the internal microphone. The processing filter includes an open-loop main filter connected at the input to the external microphone and the output of which is connected, to the adder, and to the input of the or each elementary filter.

Abstract: The headphones (10) comprise: a transducer (16) (12); an noise-canceling processing chain (30, 40) comprising: a microphone (31, 41); a noise-canceling processing filter (34, 44) which comprises in series: a stabilisation filter (34A, 44A) whose transfer function is substantially equal to the inverse of the transfer function of the processed secondary path, and a noise cancellation filter (34B, 44B) whose transfer function is a noise cancellation transfer function. The secondary path is formed between the transducer (16) and the eardrum, and the transfer function of the processed secondary path is the transfer function of the secondary path combined with the transfer functions of the processing components, excluding the processing filter (34, 44).

Abstract: The headphones (10) comprise: a transducer (16) (12); an noise-canceling processing chain (30, 40) comprising: a microphone (31, 41); a noise-canceling processing filter (34, 44) which comprises in series: a stabilisation filter (34A, 44A) whose transfer function is substantially equal to the inverse of the transfer function of the processed secondary path, and a noise cancellation filter (34B, 44B) whose transfer function is a noise cancellation transfer function. The secondary path is formed between the transducer (16) and the eardrum, and the transfer function of the processed secondary path is the transfer function of the secondary path combined with the transfer functions of the processing components, excluding the processing filter (34, 44).

Abstract: The present invention relates to a device (22) for controlling a loudspeaker in an enclosure, the device (22) comprising: a unit (36) for duplicating a desired dynamic signal (Sdyn) to obtain two identical desired dynamic signals (Sdyn1, Sdyn2), a first processing unit (38) configured to process the first desired dynamic signal (Sdyn1) to obtain a first processed signal (Sdyn1?) whereof the frequencies are less than or equal to a predetermined frequency, a second processing unit (40) configured to process the second desired dynamic signal (Sdyn2) to obtain a second processed signal (Sdyn2?) whereof the frequencies are strictly greater than the predetermined frequency, and a unit (42) for combining the processed first and second signals (Sdyn1?, Sdyn2?) to obtain a control signal (Scommande) of the loudspeaker.

Abstract: The present invention relates to a device (22) for controlling a loudspeaker in an enclosure, the device (22) comprising: a unit (36) for duplicating a desired dynamic signal (Sdyn) to obtain two identical desired dynamic signals (Sdyn1, Sdyn2), a first processing unit (38) configured to process the first desired dynamic signal (Sdyn1) to obtain a first processed signal (Sdyn1?) whereof the frequencies are less than or equal to a predetermined frequency, a second processing unit (40) configured to process the second desired dynamic signal (Sdyn2) to obtain a second processed signal (Sdyn2?) whereof the frequencies are strictly greater than the predetermined frequency, and a unit (42) for combining the processed first and second signals (Sdyn1?, Sdyn2?) to obtain a control signal (Scommande) of the loudspeaker.

Abstract: The headset includes an active noise control with an internal microphone (28) and an external microphone (32). A processor (42) comprises a feedback branch (46), adjusted so as to attenuate the low frequencies corresponding to a component of a voice signal transmitted by bone conduction, and a feedforward branch (58) adjusted so as to compensate for the attenuation introduced by the feedback filtering and the passive acoustic attenuation introduced between the outside and the inside of the headset. A voice activity detector (60) operates a dynamic switching between two couples (HFB, HFF) of different transfer functions applied to the feedback (46) and feedforward (58) functions. This allows rendering in the most natural way possible to the user all the external sounds, including his own voice.

Abstract: The headset includes an active noise control, with an internal ANC microphone (28) placed inside the acoustic cavity (22) and delivering a signal including an acoustic noise component. A digital signal processor DSP (50) comprises a feedback ANC branch (54) applying a filtering transfer function (54, HFB2) to the signal delivered by the ANC microphone, and a mixer (60) for mixing the signal of the feedback branch with an audio signal to be reproduced (M). The headset comprises a movement sensor (64) mounted on one of the earphones. The DSP comprises an anti-saturation module (68) for analyzing concurrently i) the signal delivered by the internal microphone (28) and ii) the signal delivered by the movement sensor (64), and verifying whether current characteristics of these signals fulfill or not a set of predetermined criteria. Upstream from the feedback ANC filter (54), an anti-saturation filter (70, HFB1) is selectively switched as a function of the result of this verification.

Abstract: The method comprises the determination of an observation vector that comprises only electrical measurements of the voltage (Umes) at the loudspeaker terminals and of the current (i) passing through the loudspeaker, and a state vector (X) whose components comprise: values of linear parameters of the loudspeaker response such as the electrical (Re) and mechanical (Req) resistance, and polynomial coefficients of nonlinear parameters such as the force factor (BI), the equivalent stiffness (Keq) and the electrical inductance (Le). The voltage and current measurements are applied to an estimator with a predictive filter of the extended Kalman filter incorporating a representation of a dynamic model of the loudspeaker. This filter operates a prediction of the state vector (X) and readjusts this prediction by calculation of an estimate (Uest) of the voltage based on the state vector and on the measured current and comparison of this estimate with the measurement (Umes) of the voltage.

Abstract: The headset includes an active noise control, with an internal ANC microphone (28) placed inside the acoustic cavity (22) and delivering a signal including an acoustic noise component. A digital signal processor DSP (50) comprises a feedback ANC branch (54) applying a filtering transfer function (54, HFB2) to the signal delivered by the ANC microphone, and means (46) for mixing the signal of the feedback branch with an audio signal to be reproduced (M). The headset comprises a movement sensor (64) mounted on one of the earphones. The DSP comprises means (68) for analysing concurrently i) the signal delivered by the internal microphone (28) and ii) the signal delivered by the movement sensor (64), and verifying whether current characteristics of these signals fulfil or not a set of predetermined criteria. Upstream from the feedback ANC filter (54), an anti-saturation filter (70, HFB1) is selectively switched as a function of the result of this verification.

Abstract: The headset includes an active noise control system, with an ANC microphone delivering a signal including an acoustic noise component. A digital signal processor DSP (50) comprises a feedback ANC branch (54) applying a filtering transfer function (HFB) to the signal picked up by the ANC microphone, and means (46) for mixing the signal of the feedback branch with an audio signal to be reproduced (M). The ANC microphone is an internal microphone (28) placed inside the acoustic cavity (22), and the feedback ANC filter (54) is one between a plurality of selectively switchable, pre-configured feedback ANC filter. The DSP (50) comprises means (62) for verifying whether current characteristics of the microphone signal fulfil or not a set of predetermined criteria, and for selecting one of the pre-configured feedback ANC filters as a function of the result of this verification.

Abstract: The method comprises the steps of: a) converting the digital audio signal (PCM) into a voltage signal (VE); b) first lowshelf filtering of fixed gain (G2); c) calculating a value of excursion (x) of the loudspeaker; d) comparing the excursion with a maximum value and calculating a first gain of possible attenuation (G3); e) second lowshelf filtering of gain (G4+G3) taking into account the first gain of possible attenuation; g) comparing with the maximum saturation or clipping voltage (vMAX) and calculating a second gain of possible attenuation (G5); h) third lowshelf filtering gain (G6+G3+G5) taking into account the first and/or second gains of possible attenuation; i) comparing with the maximum saturation or clipping voltage (vMAX) and applying a gain of possible overall attenuation (G7); j) optionally compensating for the nonlinearities of the loudspeaker response; and k) reversely converting the signal (Vs) into a digital audio signal (S) without dimension, for later amplification.

Abstract: The method comprises the determination of an observation vector that comprises only electrical measurements of the voltage (Umes) at the loudspeaker terminals and of the current (i) passing through the loudspeaker, and a state vector (X) whose components comprise: values of linear parameters of the loudspeaker response such as the electrical (Re) and mechanical (Req) resistance, and polynomial coefficients of nonlinear parameters such as the force factor (Bl), the equivalent stiffness (Keq) and the electrical inductance (Le). The voltage and current measurements are applied to an estimator with a predictive filter of the extended Kalman filter incorporating a representation of a dynamic model of the loudspeaker. This filter operates a prediction of the state vector (X) and readjusts this prediction by calculation of an estimate (Uest) of the voltage based on the state vector and on the measured current and comparison of this estimate with the measurement (Umes) of the voltage.

Abstract: The method comprises the steps of: a) converting the digital audio signal (PCM) into a voltage signal (VE); b) first lowshelf filtering of fixed gain (G2); c) calculating a value of excursion (x) of the loudspeaker; d) comparing the excursion with a maximum value and calculating a first gain of possible attenuation (G3); e) second lowshelf filtering of gain (G4+G3) taking into account the first gain of possible attenuation; g) comparing with the maximum saturation or clipping voltage (vMAX) and calculating a second gain of possible attenuation (G5); h) third lowshelf filtering gain (G6+G3+G5) taking into account the first and/or second gains of possible attenuation; i) comparing with the maximum saturation or clipping voltage (vMAX) and applying a gain of possible overall attenuation (G7); j) optionally compensating for the nonlinearities of the loudspeaker response; and k) reversely converting the signal (Vs) into a digital audio signal (S) without dimension, for later amplification.

Abstract: An automatic gain control system applied to a audio signal as a function of the ambient noise, the system comprising: an ambient noise estimator module suitable for establishing a current noise value estimated at least from a signal provided by a microphone; and an automatic gain control module suitable for applying to the audio signal gain of a value that is determined as a function of the current noise value received from the ambient noise estimator module. According to the invention, the ambient noise estimator module comprises an MCRA estimator suitable for establishing the current noise value from a signal provided by the microphone picking up the real noise, the echo of the music, and where appropriate speech. The system also includes a module for estimating the power of the audio signal and suitable for providing the automatic gain control module with a current power value for the audio signal.

Abstract: An automatic gain control system applied to a audio signal as a function of the ambient noise, the system comprising: an ambient noise estimator module suitable for establishing a current noise value estimated at least from a signal provided by a microphone; and an automatic gain control module suitable for applying to the audio signal gain of a value that is determined as a function of the current noise value received from the ambient noise estimator module. According to the invention, the ambient noise estimator module comprises an MCRA estimator suitable for establishing the current noise value from a signal provided by the microphone picking up the real noise, the echo of the music, and where appropriate speech. The system also includes a module for estimating the power of the audio signal and suitable for providing the automatic gain control module with a current power value for the audio signal.