Patents by Inventor Abderrahman Essebbar
Abderrahman Essebbar has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).
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Publication number: 20170341822Abstract: The invention relates to a device (100) for sealing a container having a neck, said device comprising: a deformable portion (1) intended to be at least partially arranged inside the neck of the container so as to seal it in a leak tight manner, said portion being capable of deforming elastically when inserted into the neck, said portion comprising a first end intended to be oriented towards the inside of the container, and a radiofrequency identification tag (11), said device being characterised in that it comprises at least one piezoelectric component (10) electrically connected to the tag (11) and arranged inside said deformable portion (1), within a blind recess on the first end side, generating an electric voltage caused by expansion of the deformable portion containing the piezoelectric component when said deformable portion is removed from the neck, said electric voltage being capable of causing a change of state of the radiofrequency identification tag.Type: ApplicationFiled: December 22, 2015Publication date: November 30, 2017Applicant: COMMISSARIAT A L'ENERGIE ATOMIQUE ET AUX ENERGIES ALTERNATIVESInventor: Abderrahman Essebbar
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Patent number: 8238497Abstract: A device (1) for reducing the interference in a received communication signal. The device includes: an adaptive filter (2) which uses a self-reference signal to remove a coherent and stable interfering signal from the received communication signal; a detection module (3) for detecting the existence of the target signal among the received communication signals; and a demodulation module (7) for demodulating the received communication signal when the target signal is detected. Further, this device includes: a noise-classifying device (5) which detects various interfering signals of the received communication signal and determines the classification of the detected interfering signals; and a switching module (6) for selecting the input signal for the demodulation module on the basis of the determination made by the noise-classifying device and the value when the existence of the target signal is detected.Type: GrantFiled: March 11, 2010Date of Patent: August 7, 2012Assignee: Aisin Seiki Kabushiki KaishaInventor: Abderrahman Essebbar
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Publication number: 20120027142Abstract: A device (1) for reducing the interference in a received communication signal. The device includes: an adaptive filter (2) which uses a self-reference signal to remove a coherent and stable interfering signal from the received communication signal; a detection module (3) for detecting the existence of the target signal among the received communication signals; and a demodulation module (7) for demodulating the received communication signal when the target signal is detected. Further, this device includes: a noise-classifying device (5) which detects various interfering signals of the received communication signal and determines the classification of the detected interfering signals; and a switching module (6) for selecting the input signal for the demodulation module on the basis of the determination made by the noise-classifying device and the value when the existence of the target signal is detected.Type: ApplicationFiled: March 11, 2010Publication date: February 2, 2012Applicant: AISIN SEIKI KABUSHIKI KAISHAInventor: Abderrahman Essebbar
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Patent number: 7817756Abstract: A circuit and method for suppressing interference components in a received signal are provided. The circuit includes an adaptive filter configured to process a first quantity of digital signal samples per unit time when the adaptive filter is operating in a first mode, and a second quantity of digital signal samples per unit time when operating in a second mode, wherein the first quantity is less than the second quantity. The method includes implementing an adaptive filter configured to operate in an adaptive mode and in a second mode having a reduced adaptability compared to the adaptive mode; operating the adaptive filter in the adaptive mode to process a first quantity of digital signal samples per unit time; and operating the adaptive filter in the second mode to process a second quantity of digital signal samples per unit time; wherein the first quantity is less than the second quantity.Type: GrantFiled: March 21, 2007Date of Patent: October 19, 2010Assignee: Aisin Seiki Kabushiki KaishaInventors: Tarik Aouine, Frederic Coutant, Michel Gaeta, Abderrahman Essebbar, Luc Haumonte
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Patent number: 7693712Abstract: A pre-processing system of a signal of interest in an automatic speech recognition system in a vehicle, includes an acoustic sensor to sense the signal of interest, a non acoustic sensor to sense a non acoustic noise signal, a pre-processing unit of the signal of interest, comprising a processing section of coherent frequency bands signals for suppressing the noise from the received signal, a processing section of non coherent frequency bands signals, comprising transfer function estimation device of a signal through the vehicle cabin, and a methods selection section for determining the coherence properties of the received signal, and for selecting the processing section of coherent frequency bands signals or the processing section of non coherent frequency bands signals depending on the result of the properties of the received signal.Type: GrantFiled: March 27, 2006Date of Patent: April 6, 2010Assignee: Aisin Seiki Kabushiki KaishaInventors: Michel Gaeta, Abderrahman Essebbar
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Publication number: 20080013657Abstract: A circuit and method for suppressing interference components in a received signal are provided. The circuit includes an adaptive filter configured to process a first quantity of digital signal samples per unit time when the adaptive filter is operating in a first mode, and a second quantity of digital signal samples per unit time when operating in a second mode, wherein the first quantity is less than the second quantity. The method includes implementing an adaptive filter configured to operate in an adaptive mode and in a second mode having a reduced adaptability compared to the adaptive mode; operating the adaptive filter in the adaptive mode to process a first quantity of digital signal samples per unit time; and operating the adaptive filter in the second mode to process a second quantity of digital signal samples per unit time; wherein the first quantity is less than the second quantity.Type: ApplicationFiled: March 21, 2007Publication date: January 17, 2008Applicant: AISIN SEIKI KABUSHIKI KAISHAInventors: Tarik Aouine, Frederic Coutant, Michel Gaeta, Abderrahman Essebbar, Luc Haumonte
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Publication number: 20060217977Abstract: A pre-processing system of a signal of interest in an automatic speech recognition system in a vehicle, includes an acoustic sensor to sense the signal of interest, a non acoustic sensor to sense a non acoustic noise signal, a pre-processing unit of the signal of interest, comprising a processing section of coherent frequency bands signals for suppressing the noise from the received signal, a processing section of non coherent frequency bands signals, comprising transfer function estimation device of a signal through the vehicle cabin, and a methods selection section for determining the coherence properties of the received signal, and for selecting the processing section of coherent frequency bands signals or the processing section of non coherent frequency bands signals depending on the result of the properties of the received signal.Type: ApplicationFiled: March 27, 2006Publication date: September 28, 2006Applicant: AISIN SEIKI KABUSHIKI KAISHAInventors: Michel Gaeta, Abderrahman Essebbar
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Patent number: 6704703Abstract: The excitation in a CELP-like speech coder is recursively calculated. For a given bitrate and a given complexity, the recursive approach described lowers the complexity with minimum impact on speech quality. The excitation signal is a sum of at least three vector terms, each vector term being a product of a codebook vector zk and an associated gain term gk. A first vector term g0z0 is determined that is representative of a target excitation vector x. Each remaining vector term is recursively determined as a vector term gkzk representative of the difference between the target excitation vector x and the sum of previously determined vector terms, ∑ i = 0 k - 1 ⁢ g i ⁢ z i .Type: GrantFiled: February 2, 2001Date of Patent: March 9, 2004Assignee: ScanSoft, Inc.Inventors: Mohand Ferhaoul, Jean-Francois Rasaminjanahary, Stefaan Van Gerven, Abderrahman Essebbar
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Publication number: 20010044717Abstract: The excitation in a CELP-like speech coder is recursively calculated. For a given bitrate and a given complexity, the recursive approach described lowers the complexity with minimum impact on speech quality. The excitation signal is a sum of at least three vector terms, each vector term being a product of a codebook vector zk and an associated gain term gk. A first vector term g0z0 is determined that is representative of a target excitation vector x. Each remaining vector term is recursively determined as a vector term gkzk representative of the difference between the target excitation vector x and the sum of previously determined vector terms, 1 ∑ i = 0 k - 1 ⁢ g 1 ⁢ z 1 .Type: ApplicationFiled: February 2, 2001Publication date: November 22, 2001Inventors: Mohand Ferhaoui, Jean-Francois Rasaminjanahary, Stefaan Van Gerven, Abderrahman Essebbar