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).

  • Publication number: 20170341822
    Abstract: 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: Application
    Filed: December 22, 2015
    Publication date: November 30, 2017
    Applicant: COMMISSARIAT A L'ENERGIE ATOMIQUE ET AUX ENERGIES ALTERNATIVES
    Inventor: Abderrahman Essebbar
  • Patent number: 8238497
    Abstract: 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: Grant
    Filed: March 11, 2010
    Date of Patent: August 7, 2012
    Assignee: Aisin Seiki Kabushiki Kaisha
    Inventor: Abderrahman Essebbar
  • Publication number: 20120027142
    Abstract: 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: Application
    Filed: March 11, 2010
    Publication date: February 2, 2012
    Applicant: AISIN SEIKI KABUSHIKI KAISHA
    Inventor: Abderrahman Essebbar
  • Patent number: 7817756
    Abstract: 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: Grant
    Filed: March 21, 2007
    Date of Patent: October 19, 2010
    Assignee: Aisin Seiki Kabushiki Kaisha
    Inventors: Tarik Aouine, Frederic Coutant, Michel Gaeta, Abderrahman Essebbar, Luc Haumonte
  • Patent number: 7693712
    Abstract: 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: Grant
    Filed: March 27, 2006
    Date of Patent: April 6, 2010
    Assignee: Aisin Seiki Kabushiki Kaisha
    Inventors: Michel Gaeta, Abderrahman Essebbar
  • Publication number: 20080013657
    Abstract: 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: Application
    Filed: March 21, 2007
    Publication date: January 17, 2008
    Applicant: AISIN SEIKI KABUSHIKI KAISHA
    Inventors: Tarik Aouine, Frederic Coutant, Michel Gaeta, Abderrahman Essebbar, Luc Haumonte
  • Publication number: 20060217977
    Abstract: 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: Application
    Filed: March 27, 2006
    Publication date: September 28, 2006
    Applicant: AISIN SEIKI KABUSHIKI KAISHA
    Inventors: Michel Gaeta, Abderrahman Essebbar
  • Patent number: 6704703
    Abstract: 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: Grant
    Filed: February 2, 2001
    Date of Patent: March 9, 2004
    Assignee: ScanSoft, Inc.
    Inventors: Mohand Ferhaoul, Jean-Francois Rasaminjanahary, Stefaan Van Gerven, Abderrahman Essebbar
  • Publication number: 20010044717
    Abstract: 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: Application
    Filed: February 2, 2001
    Publication date: November 22, 2001
    Inventors: Mohand Ferhaoui, Jean-Francois Rasaminjanahary, Stefaan Van Gerven, Abderrahman Essebbar