Pole-zero blocking matrix for low-delay far-field beamforming

    公开(公告)号:US11315543B2

    公开(公告)日:2022-04-26

    申请号:US16773259

    申请日:2020-01-27

    Abstract: A system performs pole-zero or IIR modeling and estimation of an inter-microphone transfer function between first and second microphones that output respective first and second microphone signals. The system includes a first adaptive FIR filter to which the first microphone signal is provided, a delay element that delays the second microphone signal by a predetermined delay amount, and a second adaptive FIR filter to which the delayed second microphone signal is provided. A first coefficient of the second adaptive FIR filter is constrained to a fixed non-zero value. The filters are jointly adapted to minimize an error signal that is a difference of the two filters outputs. The delay is small: approximately the acoustic propagation delay between the two microphones and is not determined by the environmental reverberation characteristics. The error signal may serve as a noise reference in a noise canceller, for implementing far-field beamforming with low delay.

    NONLINEAR ACOUSTIC ECHO CANCELLATION BASED ON TRANSDUCER IMPEDANCE
    2.
    发明申请
    NONLINEAR ACOUSTIC ECHO CANCELLATION BASED ON TRANSDUCER IMPEDANCE 有权
    基于传感器阻抗的非线性声学信号取消

    公开(公告)号:US20170078489A1

    公开(公告)日:2017-03-16

    申请号:US14852281

    申请日:2015-09-11

    Abstract: An acoustic echo cancellation (AEC) system within an audio playback system of an electronic device, such as a mobile phone, may calculate an estimation of an acoustic echo based on parameters describing the transducer reproducing the audio playback signals. Those parameters may include, for example, a resistance and/or inductance of the transducer and a current through and/or a voltage across the transducer. The acoustic echo cancellation system may predict, for example, a coil velocity of the transducer based on the transducer impedance. Then, an echo may be estimated using the predicted coil velocity. That estimated echo may be output to the transducer to cancel echo in the playback signal. Additionally, that estimated echo may be used to predict nonlinearities in the transducer output and an appropriate signal generated to cancel nonlinear behavior.

    Abstract translation: 诸如移动电话的电子设备的音频回放系统内的声学回声消除(AEC)系统可以基于描述重放音频回放信号的换能器的参数来计算声学回声的估计。 这些参数可以包括例如换能器的电阻和/或电感以及穿过换能器的电流和/或电压。 声学回波消除系统可以例如基于换能器阻抗预测换能器的线圈速度。 然后,可以使用预测的线圈速度来估计回波。 可以将该估计的回波输出到换能器以消除回放信号中的回波。 此外,该估计的回波可以用于预测换能器输出中的非线性以及产生的用于消除非线性行为的适当信号。

    Spatial cues from broadside detection

    公开(公告)号:US10264354B1

    公开(公告)日:2019-04-16

    申请号:US15714356

    申请日:2017-09-25

    Inventor: Khosrow Lashkari

    Abstract: Information from microphone signals from a microphone array may be used to identify persistent sources, such as televisions, radios, washing machines, or other stationary sources. Values representative of broadside conditions for each pair of microphone signals are received from the microphone array. By monitoring broadside conditions for microphone pairs, a position of a sound source may be identified. If a sound source is frequently identified with a broadside of the same microphone pair, then that sound source may be identified as a persistent noise source. When a broadside of a pair of microphones is identified with a noise source, a beamformer may be configured to decrease contribution of that pair of microphones to an audio signal formed from the microphone array.

    SPATIAL CLUES FROM BROADSIDE DETECTION
    4.
    发明申请

    公开(公告)号:US20190098399A1

    公开(公告)日:2019-03-28

    申请号:US15714356

    申请日:2017-09-25

    Inventor: Khosrow Lashkari

    Abstract: Information from microphone signals from a microphone array may be used to identify persistent sources, such as televisions, radios, washing machines, or other stationary sources. Values representative of broadside conditions for each pair of microphone signals are received from the microphone array. By monitoring broadside conditions for microphone pairs, a position of a sound source may be identified. If a sound source is frequently identified with a broadside of the same microphone pair, then that sound source may be identified as a persistent noise source. When a broadside of a pair of microphones is identified with a noise source, a beamformer may be configured to decrease contribution of that pair of microphones to an audio signal formed from the microphone array.

    Combined reference signal for acoustic echo cancellation

    公开(公告)号:US10013995B1

    公开(公告)日:2018-07-03

    申请号:US15591418

    申请日:2017-05-10

    Abstract: Acoustic echo cancellation (AEC) processing may be improved by performing echo cancellation using a combined multi-channel reference signal. Two or more reference signals, such as a left and right channel of a stereo source, may be combined and provided to an AEC block configured to receive the combined signal and perform AEC processing using the combined signal. The AEC block may include an adaptive filter that performs operations that cause pre-whitening of the combined reference signal and de-correlation of the individual channels within the combined reference signal. The pre-whitening of the signal flattens the spectrum of the combined reference signal, which may improve convergence speed of the AEC processing in cancelling the echo. The de-correlating of the signal cancels inter-channel correlation between the multiple channels, which may improve convergence speed of the AEC processing in cancelling the echo.

    Ambient-aware background noise reduction for hearing augmentation

    公开(公告)号:US11682376B1

    公开(公告)日:2023-06-20

    申请号:US17713302

    申请日:2022-04-05

    CPC classification number: G10K11/17853 G10K11/17837

    Abstract: An ambient-aware audio system reduces stationary noise and maintains dynamic environmental sound in a received input audio signal. The system includes a signal-to-noise ratio (SNR) estimator that estimates an a priori SNR and an a posteriori SNR, a gain function that uses the estimated SNRs as inputs to compute coefficients of a frequency domain noise reduction filter that uses the computed coefficients to filter a frame of the input audio signal to generate an output audio signal. The SNR estimator, gain function, and filter are configured to iterate over a plurality of frames of the input audio signal. The SNRs are estimated using the input audio signal and the output audio signal associated with one or more of the plurality of frames. The gain function is derived to minimize an expected value of differences between spectral amplitudes of the output audio signal and the input audio signal.

    POLE-ZERO BLOCKING MATRIX FOR LOW-DELAY FAR-FIELD BEAMFORMING

    公开(公告)号:US20210233509A1

    公开(公告)日:2021-07-29

    申请号:US16773259

    申请日:2020-01-27

    Abstract: A system performs pole-zero or IIR modeling and estimation of an inter-microphone transfer function between first and second microphones that output respective first and second microphone signals. The system includes a first adaptive FIR filter to which the first microphone signal is provided, a delay element that delays the second microphone signal by a predetermined delay amount, and a second adaptive FIR filter to which the delayed second microphone signal is provided. A first coefficient of the second adaptive FIR filter is constrained to a fixed non-zero value. The filters are jointly adapted to minimize an error signal that is a difference of the two filters outputs. The delay is small: approximately the acoustic propagation delay between the two microphones and is not determined by the environmental reverberation characteristics. The error signal may serve as a noise reference in a noise canceller, for implementing far-field beamforming with low delay.

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