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公开(公告)号:US10770076B2
公开(公告)日:2020-09-08
申请号:US16020406
申请日:2018-06-27
Inventor: César Alonso , John Paul Lesso
IPC: G10L17/10 , G10L25/03 , H04L9/32 , G06F21/32 , H04W12/12 , H04R1/40 , G10L25/78 , G07C9/37 , G10L17/22 , H04R3/00 , H04L29/06 , H04K1/00 , G06F21/55 , G10L21/0216
Abstract: A method of detecting a replay attack on a voice biometrics system comprises: receiving an audio signal representing speech; detecting a magnetic field; determining if there is a correlation between the audio signal and the magnetic field; and if there is a correlation between the audio signal and the magnetic field, determining that the audio signal may result from a replay attack.
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公开(公告)号:US10726849B2
公开(公告)日:2020-07-28
申请号:US15666280
申请日:2017-08-01
Inventor: John Paul Lesso , John Laurence Melanson
Abstract: This application describes methods and apparatus for speaker recognition. An apparatus according to an embodiment has an analyzer (202) for analyzing each frame of a sequence of frames of audio data (AIN) which correspond to speech sounds uttered by a user to determine at least one characteristic of the speech sound of that frame. An assessment module (203) determines, for each frame of audio data, a contribution indicator of the extent to which the frame of audio data should be used for speaker recognition processing based on the determined characteristic of the speech sound. In this way frames which correspond to speech sounds that are of most use for speaker discrimination may be emphasized and/or frames which correspond to speech sounds that are of least use for speaker discrimination may be de-emphasized.
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公开(公告)号:US10720171B1
公开(公告)日:2020-07-21
申请号:US16280288
申请日:2019-02-20
Inventor: John Paul Lesso , Gordon Richard McLeod
IPC: G06F17/00 , G10L19/16 , G10L19/005 , G10L21/038
Abstract: In a method of audio processing, a plurality of audio samples are received, and are concatenated to form a composite audio signal. The composite audio signal is analysed to identify audio artefacts associated with concatenation in the composite audio signal, and any identified audio artefacts are compensated for, to form a corrected composite audio signal. The corrected composite audio signal is provided to a voice biometrics module.
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公开(公告)号:US10649732B2
公开(公告)日:2020-05-12
申请号:US16050990
申请日:2018-07-31
Inventor: John Paul Lesso , Mark McCloy-Stevens
IPC: G06F7/533 , H03K19/17704
Abstract: This application relates to apparatus and methods for the multiplication of signals. A multiplication circuit (100) has first and second time-encoding modulators (103a, 103b) configured to receive first and second combined signals (SC1, SC2) respectively, and generate respective first and second PWM signals (SPWM1, SPWM2), each with a cycle frequency that depends substantially on the square of the value of the input combined signal. The first combined signal (SC1) corresponds to a sum of a first and second input signals (S1, S2) and the second combined signal (SC2) corresponds to the difference between the first and second input signals (S1, S2). First and second time-decoding converters (104a, 104b) receive the first and second PWM signals and provide respective first and count values (D1, D2) based on a parameter related to the frequency of the respective first or second PWM signal. A subtractor (105) determine a difference between the first and second count values (D1, D2) and provides an output signal (DOUT) based on this difference.
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公开(公告)号:US10587232B2
公开(公告)日:2020-03-10
申请号:US15982237
申请日:2018-05-17
Inventor: John Paul Lesso , Toru Ido
Abstract: The present disclosure relates to Class D amplifier circuitry comprising: an input for receiving an input signal; first and second output nodes for driving a load connected between the first and second output nodes. A first driver stage is provided for switching the first node between a first supply rail and a second supply rail, and a second driver stage is provided for switching the second node between the first supply rail and the second supply rail. The Class D amplifier circuitry also includes first driver control circuitry configured to receive a first carrier wave and control the switching of the first driver stage based in part on the first carrier wave; second driver control circuitry configured to receive a second carrier wave and control the switching of the second driver stage based in part on the second carrier wave; and a carrier wave generator configured to provide the first carrier wave and the second carrier wave. A phase shift between the first carrier wave and the second carrier wave is adjustable responsive to a mode control signal.
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公开(公告)号:US10554189B2
公开(公告)日:2020-02-04
申请号:US15992847
申请日:2018-05-30
Inventor: John Paul Lesso , Seyed Hamed Sadati , Yu Tamura
Abstract: This application relates to audio circuits, such as audio driving circuits, with improved audio performance. An audio arrangement (200) has an audio circuit (201) with a forward signal path between an input (102) for an input digital audio signal (DIN) and an output (103) for an output analogue audio signal (AOUT). The circuit also has a feedback path comprising an analogue-to-digital conversion module (202) for receiving an analogue feedback signal (VFB) derived from the output analogue audio signal and outputting a corresponding digital feedback signal (DFB). The analogue-to-digital conversion module (202) has an ADC (108), an analogue gain element (203) configured to apply analogue gain (GA) to the analogue feedback signal before the ADC and a digital gain element (204) for applying digital gain (GD) to a signal output from the ADC. A gain controller (205) controls the analogue gain and the digital gain applied based on the input digital audio signal (DIN).
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公开(公告)号:US10540978B2
公开(公告)日:2020-01-21
申请号:US15992562
申请日:2018-05-30
Inventor: John Paul Lesso , Gordon Richard McLeod
Abstract: A method of speaker verification comprises: comparing a test input against a model of a user's speech obtained during a process of enrolling the user; obtaining a first score from comparing the test input against the model of the user's speech; comparing the test input against a first plurality of models of speech obtained from a first plurality of other speakers respectively; obtaining a plurality of cohort scores from comparing the test input against the plurality of models of speech obtained from a plurality of other speakers; obtaining statistics describing the plurality of cohort scores; modifying said statistics to obtain adjusted statistics; normalising the first score using the adjusted statistics to obtain a normalised score; and using the normalised score for speaker verification.
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公开(公告)号:US10431212B2
公开(公告)日:2019-10-01
申请号:US15614093
申请日:2017-06-05
Inventor: John Paul Lesso , Robert James Hatfield
Abstract: A speech recognition system comprises: an input, for receiving an input signal from at least one microphone; a first buffer, for storing the input signal; a noise reduction block, for receiving the input signal and generating a noise reduced input signal; a speech recognition engine, for receiving either the input signal output from the first buffer or the noise reduced input signal from the noise reduction block; and a selection circuit for directing either the input signal output from the first buffer or the noise reduced input signal from the noise reduction block to the speech recognition engine.
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公开(公告)号:US10297258B2
公开(公告)日:2019-05-21
申请号:US15538619
申请日:2015-12-22
Inventor: John Paul Lesso , John Laurence Melanson
Abstract: A microphone unit has a transducer, for generating an electrical audio signal from a received acoustic signal; a speech coder, for obtaining compressed speech data from the audio signal; and a digital output, for supplying digital signals representing said compressed speech data. The speech coder may be a lossy speech coder, and may contain a bank of filters with center frequencies that are non-uniformly spaced, for example mel frequencies.
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公开(公告)号:US10237668B2
公开(公告)日:2019-03-19
申请号:US15935542
申请日:2018-03-26
Inventor: John Paul Lesso , Emmanuel Philippe Christian Hardy
Abstract: This application describes an apparatus (300) for monitoring for blockage of an acoustic (110) port of a microphone device (100). The apparatus has a spectrum peak detect block (301) for receiving a microphone signal (SMIC) and determining, from the microphone signal, a resonance frequency (fH) and a quality factor (QH) of a resonance (202) associated with the acoustic port. A condition monitoring block (302) is configured to determine any change in resonance frequency and quality factor and to determine a blockage status for the microphone based on said detect changes. The condition monitoring block identifies a change in blockage status if there is a change in quality factor.
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