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公开(公告)号：US11631390B2
公开(公告)日：2023-04-18
申请号：US16236861
申请日：2018-12-31
Inventor： Dayong Zhou , John L. Melanson
IPC: G10K11/178 , H04M1/60 , H04M1/03
Abstract: The overall performance of an ANC system may be improved by configuring the ANC system to perform adaption in the frequency domain. The ANC systems may be configured to update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter. Updating may include changing parameters of the algorithm based on a SDR based, at least in part, on the first input signal. Updating may also include normalizing a step size and processing at least full band information for the input signal in a frequency domain to generate coefficient values for the algorithm. Updating may also include applying a frequency domain magnitude constraint on adaptive filter coefficients.
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公开(公告)号：US11477569B2
公开(公告)日：2022-10-18
申请号：US17205352
申请日：2021-03-18
Inventor： Dayong Zhou , Brad Zwernemann
Abstract: A method of obtaining a directional microphone signal, the method comprising: receiving first and second microphone signals from first and second microphones separated by a distance; obtaining a combined microphone signal based on one or more of the first and second microphone signals; obtaining a difference microphone signal by subtracting the second microphone signal from the first microphone signal; obtaining a transformed combined microphone signal by applying a Hilbert transform to the combined microphone signal; combining the transformed combined microphone signal with the difference microphone signal to obtain the directional microphone signal.
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公开(公告)号：US20210020157A1
公开(公告)日：2021-01-21
申请号：US16516644
申请日：2019-07-19
Inventor： Dayong Zhou , Chin Huang Yong
IPC: G10K11/178
Abstract: An adaptive filter includes a frequency domain adaptation block that analyzes a statistic of coefficient movement in the frequency domain. The adaption block adjusts, in the frequency domain, a parameter (step size or leakage factor) that affects speed of convergence of the adaptive filter based on the analyzed statistic of filter coefficient movement. The filter includes an associated coefficient, statistic of coefficient movement, and parameter for each frequency bin. The coefficients may be complex numbers, and separate real and imaginary statistics and parameters are maintained. The statistic may be direction counts of the filter coefficient movement. The step size may be adjusted to a predetermined minimum value when the current direction of movement of the filter coefficient is different than the predominant direction and otherwise the step size is adjusted approximately proportionally to an amount of predominance by a value based on a direction count of the filter coefficient movement.
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公开(公告)号：US10276145B2
公开(公告)日：2019-04-30
申请号：US15495714
申请日：2017-04-24
Inventor： Dayong Zhou , Yang Lu
IPC: G10K11/16 , A61F11/06 , G10K11/178 , H04W88/02
Abstract: The handling of disturbances to audio signals may be improved with an adaptive noise cancellation (ANC) system that performs frequency-domain adaption. The ANC systems may be configured to determine if a disturbance is present at a first frequency in the second input signal received from the reference microphone. The ANC systems may update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter by changing parameters of the algorithm such that the adaptive filter adapts around the first frequency differently than other frequencies when the disturbance is present.
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公开(公告)号：US20180308466A1
公开(公告)日：2018-10-25
申请号：US15495746
申请日：2017-04-24
Inventor： Dayong Zhou , John L. Melanson
IPC: G10K11/178
Abstract: The overall performance of an ANC system may be improved by configuring the ANC system to perform adaption in the frequency domain. The ANC systems may be configured to update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter. Updating may include changing parameters of the algorithm based on a SDR based, at least in part, on the first input signal. Updating may also include normalizing a step size and processing at least full band information for the input signal in a frequency domain to generate coefficient values for the algorithm. Updating may also include applying a frequency domain magnitude constraint on adaptive filter coefficients.
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Patent Agency Ranking
6.发明申请HYBRID ADAPTIVE NOISE CANCELLATION SYSTEM WITH FILTERED ERROR MICROPHONE SIGNAL 有权混合自适应噪声消除系统与过滤错误麦克风信号公开(公告)号：US20170053638A1
公开(公告)日：2017-02-23
申请号：US14832585
申请日：2015-08-21
Inventor： Dayong Zhou , Yang Lu , Ning Li , Nitin Kwatra , Antonio J. Miller
IPC: G10K11/178
CPC classification number: G10K11/178 , G10K11/16 , G10K11/17817 , G10K11/17879 , G10K11/17885 , G10K2210/108 , G10K2210/1081 , G10K2210/1082 , G10K2210/3017 , G10K2210/3022 , G10K2210/3026 , G10K2210/3027 , G10K2210/3035 , G10K2210/3039 , G10K2210/3045 , G10K2210/3049 , G10K2210/3053 , G10K2210/3055 , G10K2210/3056 , G10K2210/509 , H04R1/1083 , H04R3/002 , H04R3/005 , H04R2410/05 , H04R2460/01 , H04R2499/11
Abstract: In accordance with systems and methods of the present disclosure, an adaptive noise cancellation system may include an alignment filter configured to correct misalignment of a reference microphone signal and an error microphone signal by generating a misalignment correction signal.
Abstract translation: 根据本公开的系统和方法，自适应噪声消除系统可以包括对准滤波器，其被配置为通过产生未对准校正信号来校正参考麦克风信号和误差麦克风信号的未对准。
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公开(公告)号：US12149899B2
公开(公告)日：2024-11-19
申请号：US17847319
申请日：2022-06-23
Inventor： Dayong Zhou , Brad Zwernemann , Kaichow Lau , Dana J. Taipale , John L. Melanson
IPC: H04R3/02
Abstract: Circuitry for acoustic crosstalk cancellation between first and second acoustic signals, the circuitry comprising: crosstalk cancellation circuitry configured to: receive a first audio signal and, based on the received first audio signal, generate a first crosstalk cancellation signal; receive a second audio signal and, based on the received second audio signal, generate a second crosstalk cancellation signal; combine the first crosstalk cancellation signal with a signal indicative of the second audio signal to generate a first crosstalk cancellation circuitry output signal; and combine the second crosstalk cancellation signal with a signal indicative of the first audio signal to generate a second crosstalk cancellation circuitry output signal; and output stage circuitry configured to: receive the first crosstalk cancellation circuitry output signal and, based on the received first crosstalk cancellation circuitry, generate a first drive signal for driving a first speaker to generate the first acoustic signal; and receive the second crosstalk cancellation circuitry output signal and, based on the received second crosstalk cancellation circuitry, generate a second drive signal for driving a second speaker to generate the second acoustic signal, wherein a parameter of the crosstalk cancellation circuitry is variable based on one or more of: a position of a user of a host device incorporating the circuitry with respect to the host device; a volume setting of the host device; a level of the first and/or second crosstalk cancellation signal; and an operational parameter of the output stage circuitry.
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公开(公告)号：US11468873B2
公开(公告)日：2022-10-11
申请号：US15720407
申请日：2017-09-29
Inventor： Sang-Ik Cho , Dayong Zhou
IPC: G10K11/178
Abstract: An integrated circuit for implementing at least a portion of a personal audio device may include a processing circuit to implement an adaptive filter having a response that generates an anti-noise signal to reduce the presence of the ambient audio sounds at an error microphone, implement a coefficient control block that shapes the response of the adaptive filter in conformity with the error microphone signal by computing coefficients that determine the response of the adaptive filter to minimize the ambient audio sounds at the error microphone, and responsive to detecting a condition that triggers a reset of the adaptive filter, increment the coefficients in a plurality of steps from initial values of the coefficients at a time of triggering the reset to final values of the coefficients at a conclusion of the reset.
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公开(公告)号：US10984778B2
公开(公告)日：2021-04-20
申请号：US16516644
申请日：2019-07-19
Inventor： Dayong Zhou , Chin Huang Yong
IPC: G10K11/178
Abstract: An adaptive filter includes a frequency domain adaptation block that analyzes a statistic of coefficient movement in the frequency domain. The adaption block adjusts, in the frequency domain, a parameter (step size or leakage factor) that affects speed of convergence of the adaptive filter based on the analyzed statistic of filter coefficient movement. The filter includes an associated coefficient, statistic of coefficient movement, and parameter for each frequency bin. The coefficients may be complex numbers, and separate real and imaginary statistics and parameters are maintained. The statistic may be direction counts of the filter coefficient movement. The step size may be adjusted to a predetermined minimum value when the current direction of movement of the filter coefficient is different than the predominant direction and otherwise the step size is adjusted approximately proportionally to an amount of predominance by a value based on a direction count of the filter coefficient movement.
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公开(公告)号：US10789933B1
公开(公告)日：2020-09-29
申请号：US16516618
申请日：2019-07-19
Inventor： Ning Li , Dayong Zhou
IPC: G10K11/178
Abstract: An adaptive filter calculates frequency domain coefficients and in the frequency domain dynamically adjusts a leakage/step size parameter that controls adaptation of the adaptive filter based on the calculated frequency domain coefficients (e.g., based on a peak magnitude of the coefficients among frequency bins or on the magnitude of the coefficient of the corresponding frequency bin). The adaptive filter calculates the coefficients based on frequency domain input and error signals, dynamically adjusts a frequency domain coefficient magnitude limit parameter based on the calculated frequency domain coefficients (e.g., approximately proportionally to a peak magnitude of the coefficients among frequency bins) and uses the dynamically adjusted frequency domain coefficient magnitude limit parameter to limit a magnitude of the calculated frequency domain coefficients. The limit may be engaged above a frequency bin based on the peak magnitude frequency bin. An ANC system may employ the filter.
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