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    Sound processing in a hearing device using externally and internally received sounds
    62.
    发明授权
    Sound processing in a hearing device using externally and internally received sounds 有权

    公开(公告)号：US09716952B2

    公开(公告)日：2017-07-25

    申请号：US14863686

    申请日：2015-09-24

    申请人： Stefan Mauger

    发明人： Stefan Mauger

    IPC分类号： H04R25/00 ,  A61N1/36 ,  G10L21/0264 ,  G10L21/0216

    CPC分类号： H04R25/505 ,  A61N1/36036 ,  G10L21/0216 ,  G10L21/0264 ,  G10L2021/02165 ,  H04R25/43 ,  H04R2225/43 ,  H04R2225/67 ,  H04R2410/07

    摘要： Disclosed herein are methods, systems, and devices for mitigating the impact of noise, such a low-frequency noise caused by wind, on sounds received at a hearing prosthesis. An example method includes receiving an external sound signal transduced externally to a recipient from an ambient sound and an internal sound signal transduced internally to the recipient from the ambient sound. The example method also includes determining that a triggering condition is present in the external sound signal. The triggering condition may be indicative of a condition in that more adversely affects externally-received sounds than internally-received sounds. In response to determining that the triggering condition is present in the external sound signal, the example method further includes generating a stimulation signal that is based at least in part on spectral information of the internally-transduced sound.

    ACTIVE NOISE-CONTROL SYSTEM WITH SOURCE-SEPARATED REFERENCE SIGNAL
    63.
    发明申请
    ACTIVE NOISE-CONTROL SYSTEM WITH SOURCE-SEPARATED REFERENCE SIGNAL 有权

    公开(公告)号：US20170193975A1

    公开(公告)日：2017-07-06

    申请号：US14986527

    申请日：2015-12-31

    申请人： Harman International Industries, Inc.

    发明人： Donald Joseph Butts

    IPC分类号： G10K11/178

    CPC分类号： G10K11/178 ,  F01N1/065 ,  F01N2560/14 ,  G10K2210/3012 ,  G10K2210/3027 ,  G10K2210/3028 ,  G10L21/0216 ,  G10L21/0272

    摘要： The various embodiments set forth an active noise cancellation system that includes a source separation algorithm. The source separation algorithm enables the identification of acoustic inputs from a particular sound source based on a reference signal generated with one or more microphones. Consequently, the identified acoustic inputs can be cancelled or damped in a targeted listening location via an acoustic correction signal, where the acoustic correction signal is generated based on a sound source separated from the reference signal. Advantageously, the reference signal can be generated with a microphone, even though such a reference signal may include a combination of multiple acoustic inputs. Thus, noise sources that cannot be individually measured, for example with an accelerometer mounted to a vibrating structure, can still be identified and actively cancelled.

    ADAPTIVE BEAMFORMING TO CREATE REFERENCE CHANNELS
    65.
    发明申请
    ADAPTIVE BEAMFORMING TO CREATE REFERENCE CHANNELS 有权

    公开(公告)号：US20170178662A1

    公开(公告)日：2017-06-22

    申请号：US14973274

    申请日：2015-12-17

    申请人： Amazon Technologies, Inc.

    发明人： Robert Ayrapetian ,  Philip Ryan Hilmes

    IPC分类号： G10L21/0216 ,  H04R5/04

    CPC分类号： G10L21/0216 ,  G10L21/0208 ,  G10L2021/02082 ,  G10L2021/02166 ,  H04R3/005 ,  H04R5/04 ,  H04R2201/40 ,  H04R2203/12 ,  H04R2420/07

    摘要： An echo cancellation system that performs audio beamforming to separate audio input into multiple directions and determines a target signal and a reference signal from the multiple directions. For example, the system may detect a strong signal associated with a speaker and select the strong signal as a reference signal, selecting another direction as a target signal. The system may determine a speech position and may select the speech position as a target signal and an opposite direction as a reference signal. The system may create pairwise combinations of opposite directions, with an individual direction being selected as a target signal and a reference signal. The system may select a fixed beamformer output for the target signal and an adaptive beamformer output for the reference signal, or vice versa. The system may remove the reference signal (e.g., audio output by the loudspeaker) to isolate speech included in the target signal.

    Method and device for suppressing residual echoes based on inverse transmitter receiver distance and delay for speech signals directly incident on a transmitter array
    66.
    发明授权
    Method and device for suppressing residual echoes based on inverse transmitter receiver distance and delay for speech signals directly incident on a transmitter array 有权

    公开(公告)号：US09685172B2

    公开(公告)日：2017-06-20

    申请号：US13642661

    申请日：2011-10-24

    申请人： Shasha Lou ,  Bo Li ,  Song Liu

    发明人： Shasha Lou ,  Bo Li ,  Song Liu

    IPC分类号： H04M9/08 ,  G10L15/00 ,  G10L21/0216 ,  G10L15/20 ,  H04R3/00 ,  G10L21/0208

    CPC分类号： G10L21/0216 ,  G10L15/20 ,  G10L2021/02082 ,  H04M9/082 ,  H04R3/005 ,  H04R2201/109 ,  H04R2499/13

    摘要： The present invention discloses a method and a device for suppressing residual echoes. The method comprises: performing adaptive filtering on M transmitter signals respectively to obtain M adaptive filtered signals; performing array-filtering 5 on the M−1 adaptive filtered signals other than the first adaptive filtered signal to obtain M−1 array-filter output signals by considering relative positions of the receiver and each of the transmitters and the time delay attributed to distances between the transmitters and the receiver; subtracting each of the M−1 array-filter output signals from the first adaptive filtered signal respectively to obtain M−1 difference signals, performing time-domain/frequency-domain conversion on the M−1 difference signals respectively and selecting one of the frequency-domain signals that has the least energy; performing time-domain/frequency-domain conversion on the first adaptive filtered signal and the Mth adaptive filtered signal and then performing speech probability filtering on the converted first adaptive filtered signal and the converted Mth adaptive filtered signal to obtain one frequency-domain speech probability signal; and multiplying the frequency-domain speech probability signal with the selected signal that has the least energy, and performing frequency-domain/time-domain conversion on the multiplication result to obtain a signal as a transmitter output signal. The technical solutions of the present invention can suppress the residual echoes effectively without impairing near end speech.

    Microphone Signal Fusion
    68.
    发明申请
    Microphone Signal Fusion 有权
    麦克风信号融合

    公开(公告)号：US20170078790A1

    公开(公告)日：2017-03-16

    申请号：US15213203

    申请日：2016-07-18

    申请人： Knowles Electronics, LLC

    发明人： Kuan-Chieh Yen ,  Thomas E. Miller ,  Mushtaq Syed

    IPC分类号： H04R3/00 ,  G10L21/0232 ,  G10L21/0308

    CPC分类号： H04R3/005 ,  G10L21/0216 ,  G10L21/0232 ,  G10L21/0308 ,  G10L2021/02165 ,  G10L2021/02166 ,  H04R1/1016 ,  H04R1/1041 ,  H04R1/1083 ,  H04R1/406 ,  H04R2201/107 ,  H04R2225/43 ,  H04R2410/05 ,  H04R2420/07 ,  H04R2430/03 ,  H04R2460/13 ,  H04R2499/11

    摘要： Provided are systems and methods for microphone signal fusion. An example method commences with receiving a first and second signal representing sounds captured, respectively, by external and internal microphones. The internal microphone is located inside an ear canal and sealed for isolation from outside acoustic signals. The external microphone is located outside the ear canal. The first signal comprises a voice component. The second signal comprises a voice component modified by at least human tissue. The first and second signals are processed to obtain noise estimates. The voice component of the second signal is aligned with the voice component of the first signal. The first signal and the aligned voice component of the second signal are blended, based on the noise estimates, to generate an enhanced voice signal. Prior to aligning, the voice component of the second signal may be processed to emphasize high frequency content, improving effective alignment bandwidth.

    摘要翻译： 提供了用于麦克风信号融合的系统和方法。 示例性方法开始于分别由外​​部和内部麦克风接收表示所捕获的声音的第一和第二信号。 内置麦克风位于耳道内部，并密封以隔离外部声学信号。 外部麦克风位于耳道外。 第一信号包括语音分量。 第二信号包括由至少人体组织修改的声音成分。 处理第一和第二信号以获得噪声估计。 第二信号的声音分量与第一信号的声音分量对齐。 基于噪声估计，混合第二信号的第一信号和对准的声音分量，以产生增强的声音信号。 在对齐之前，可以处理第二信号的语音分量以强调高频内容，改善有效的对准带宽。

    Accurate Forward SNR Estimation Based on MMSE Speech Probability Presence
    69.
    发明申请
    Accurate Forward SNR Estimation Based on MMSE Speech Probability Presence 有权
    基于MMSE语音概率准确的前向SNR估计

    公开(公告)号：US20170004842A1

    公开(公告)日：2017-01-05

    申请号：US15269357

    申请日：2016-09-19

    申请人： Continental Automotive Systems, Inc.

    发明人： Guillaume Lamy ,  Bijal Joshi

    IPC分类号： G10L21/0232 ,  G10L25/21 ,  G10L25/84

    CPC分类号： G10L21/0232 ,  G10L21/0208 ,  G10L21/0216 ,  G10L25/21 ,  G10L25/84

    摘要： Acoustic noise in an audio signal is reduced by calculating a speech probability presence (SPP) factor using minimum mean square error (MMSE). The SPP factor, which has a value typically ranging between zero and one, is modified or warped responsive to a value obtained from the evaluation of a sigmoid function, the shape of which is determined by a signal-to-noise ratio (SNR), which is obtained by an evaluation of the signal energy and noise energy output from a microphone over time. The shape and aggressiveness of the sigmoid function is determined using an extrinsically-determined SNR, not determined by the MMSE determination. The extrinsically-determined SNR is obtained from a long term history of previously-determined speech presence probabilities and a long term history of previously-determined noise histories.

    摘要翻译： 通过使用最小均方误差（MMSE）计算语音概率存在（SPP）因子来减少音频信号中的声音噪声。 具有通常在零和1之间的值的SPP因子响应于从S估计获得的值而修改或变形，该S值的形状由信噪比（SNR）确定， 这是通过评估从麦克风输出的信号能量和噪声能量随时间而获得的。 使用不是由MMSE确定确定的外部确定的SNR来确定S形函数的形状和侵略性。 从外部确定的SNR是从先前确定的语音存在概率的长期历史和先前确定的噪声历史的长期历史获得的。

    NOISE CANCELATION SYSTEM AND TECHNIQUES
    70.
    发明申请
    NOISE CANCELATION SYSTEM AND TECHNIQUES 有权
    噪声消除系统和技术

    公开(公告)号：US20170004818A1

    公开(公告)日：2017-01-05

    申请号：US14789298

    申请日：2015-07-01

    申请人： zPillow, Inc.

    发明人： Chidananda KHATUA

    IPC分类号： G10K11/178 ,  H04R1/02 ,  H04R29/00

    CPC分类号： G10K11/178 ,  G10K11/17857 ,  G10K11/17875 ,  G10K2210/12 ,  G10K2210/3012 ,  G10K2210/3215 ,  G10K2210/3221 ,  G10L21/0216 ,  H04R1/028 ,  H04R1/406 ,  H04R29/005 ,  H04R2203/12 ,  H04R2410/01 ,  H04R2410/05 ,  H04R2430/20

    摘要： Techniques for noise cancelation include an automated method having the steps of: receiving signals from a plurality of microphones positioned within a microphone array outside a target area; identifying, from the received signals, a noise and position information for a source for the noise external to the target area before the noise reaches the target area; before the noise reaches the target area, determining a cancelation sound for the noise based on the noise and the position information; and playing the cancelation sound as the noise reaches the target area so as to significantly cancel the noise within the target area.

    摘要翻译： 用于噪声消除的技术包括自动化方法，其具有以下步骤：从位于目标区域外的麦克风阵列内的多个麦克风接收信号; 在所述噪声到达所述目标区域之前，从所接收的信号中识别用于所述目标区域外部的噪声的源的噪声和位置信息; 在噪声到达目标区域之前，基于噪声和位置信息确定噪声的消除声音; 并且当噪声到达目标区域时播放消除声音，以便显着地消除目标区域内的噪声。