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1. 发明授权

US4130726A Loudspeaker system equalization 失效
标题翻译：扬声器系统均衡
公开(公告)号：US4130726A
公开(公告)日：1978-12-19
申请号：US811297
申请日：1977-06-29
申请人： James M. Kates , Robert A. Berkovitz
发明人： James M. Kates , Robert A. Berkovitz
IPC分类号： H03G5/02 , H04R3/04
CPC分类号： H04R3/04 , H03G5/025
摘要： Variations in the amplitude-frequency response characteristic of a loudspeaker are compensated by a drive circuit in which an audio input signal is summed with a plurality of delayed replicas of the input signal to provide a drive signal for the loudspeaker. The delayed replicas are derived by discrete time filtering. Noise reduction techniques applied to the least delayed of the plurality of delay signals substantially improves the S/N ratio in the drive signal.
摘要翻译：扬声器的振幅 - 频率响应特性的变化由驱动电路补偿，其中音频输入信号与输入信号的多个延迟复制相加，以提供扬声器的驱动信号。延迟复制品是通过离散时间滤波得出的。应用于多个延迟信号的最小延迟的降噪技术大大改善了驱动信号中的S / N比。

2. 发明授权

US08036404B2 Binaural signal enhancement system 有权
标题翻译：双耳信号增强系统
公开(公告)号：US08036404B2
公开(公告)日：2011-10-11
申请号：US12029292
申请日：2008-02-11
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H04R25/00
CPC分类号： H04R25/407 , H04R25/552 , H04R2225/41
摘要： A signal processing system, such as a hearing aid system, adapted to enhance binaural input signals is provided. The signal processing system is essentially a system with a first signal channel having a first filter and a second signal channel having a second filter for processing first and second channel inputs and producing first and second channel outputs, respectively. Filter coefficients of at least one of the first and second filters are adjusted to minimize the difference between the first channel input and the second channel input in producing the first and second channel outputs. The resultant signal match processing of the signal processing system gives broader regions of signal suppression than using the Wiener filters alone for frequency regions where the interaural correlation is low, and may be more effective in reducing the effects of interference on the desired speech signal. Modifications to the algorithms can be made to accommodate sound sources located to the sides as well as the front of the listener. Processing artifacts can be reduced by using longer averaging time constants for estimating the signal power and cross-spectra as the signal-to-noise ratio decreases. A stability constant can also be incorporated in the transfer functions of the first and second filters to increase the stability of the signal processing system.
摘要翻译：提供了一种适于增强双耳输入信号的信号处理系统，例如助听器系统。信号处理系统本质上是具有第一信号信道的系统，其具有第一滤波器和具有用于处理第一和第二信道输入的第二滤波器的第二信号信道，并且分别产生第一和第二信道输出。调整第一和第二滤波器中的至少一个滤波器的滤波器系数，以便在产生第一和第二通道输出时最小化第一通道输入和第二通道输入之间的差异。信号处理系统的结果信号匹配处理给出了比单独使用维纳滤波器用于频偏相关低的频率区域的更宽的信号抑制区域，并且可以更有效地减少干扰对期望语音信号的影响。可以对算法进行修改以适应位于侧面以及收听者前面的声源。通过使用较长的平均时间常数来减少处理伪影，以估计信噪比降低信号功率和交叉谱。稳定常数也可以并入第一和第二滤波器的传递函数中，以增加信号处理系统的稳定性。

3. 发明授权

US07630507B2 Binaural compression system 有权
标题翻译：双耳压缩系统
公开(公告)号：US07630507B2
公开(公告)日：2009-12-08
申请号：US10353187
申请日：2003-01-27
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H04R25/00
CPC分类号： H04B1/64 , H03G7/06 , H04R25/356 , H04R25/552
摘要： A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs. The multi-channel signal processing system further includes peak detecting means detecting signal peaks of the first and second channel input signals for generating first and second channel control signals. Thereafter, gain adjusting means adjusts signal gains of the first and second channel control signals. The first and second channel compressing units then respectively compress the first and second channel input signals to produce the first and second channel compressed outputs in accordance with the adjusted first and second channel control signals, respectively.
摘要翻译：提供了一种适用于提供音调输入的双耳压缩的多声道信号处理系统。这样的系统可以用于例如双耳助听器系统中以提供音调输入的动态范围双耳压缩。多通道信号处理系统本质上是一种具有通过两个信号通道之间的控制链路连接的两个信号通道的系统，从而允许双耳助听器系统模拟诸如人类的交叉橄榄色丛（COCB）效应的行为听觉系统包括左耳和右耳之间的神经连接。多信道信号处理系统包括分别位于多信道信号处理系统的第一和第二信号信道中的第一和第二信道压缩单元。第一和第二信道压缩单元分别接收第一和第二信道输入信号，以产生第一和第二信道压缩输出。多通道信号处理系统还包括检测第一和第二信道输入信号的信号峰值的峰值检测装置，用于产生第一和第二信道控制信号。此后，增益调整装置调整第一和第二信道控制信号的信号增益。第一和第二信道压缩单元然后分别压缩第一和第二信道输入信号，以分别根据调整的第一和第二信道控制信号产生第一和第二信道压缩输出。

4. 发明授权

US07330556B2 Binaural signal enhancement system 失效
标题翻译：双耳信号增强系统
公开(公告)号：US07330556B2
公开(公告)日：2008-02-12
申请号：US10407305
申请日：2003-04-03
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H04R25/00
CPC分类号： H04R25/407 , H04R25/552 , H04R2225/41
摘要： A signal processing system, such as a hearing aid system, adapted to enhance binaural input signals is provided. The signal processing system is essentially a system with a first signal channel having a first filter and a second signal channel having a second filter for processing first and second channel inputs and producing first and second channel outputs, respectively. Filter coefficients of at least one of the first and second filters are adjusted to minimize the difference between the first channel input and the second channel input in producing the first and second channel outputs. The resultant signal match processing of the signal processing system gives broader regions of signal suppression than using the Wiener filters alone for frequency regions where the interaural correlation is low, and may be more effective in reducing the effects of interference on the desired speech signal. Modifications to the algorithms can be made to accommodate sound sources located to the sides as well as the front of the listener. Processing artifacts can be reduced by using longer averaging time constants for estimating the signal power and cross-spectra as the signal-to-noise ratio decreases. A stability constant can also be incorporated in the transfer functions of the first and second filters to increase the stability of the signal processing system.
摘要翻译：提供了一种适于增强双耳输入信号的信号处理系统，例如助听器系统。信号处理系统本质上是具有第一信号信道的系统，其具有第一滤波器和具有用于处理第一和第二信道输入的第二滤波器的第二信号信道，并且分别产生第一和第二信道输出。调整第一和第二滤波器中的至少一个滤波器的滤波器系数，以便在产生第一和第二通道输出时最小化第一通道输入和第二通道输入之间的差异。信号处理系统的结果信号匹配处理给出了比单独使用维纳滤波器用于频偏相关低的频率区域的更宽的信号抑制区域，并且可以更有效地减少干扰对期望语音信号的影响。可以对算法进行修改以适应位于侧面以及收听者前面的声源。通过使用较长的平均时间常数来减少处理伪影，以估计信噪比降低信号功率和交叉谱。稳定常数也可以并入第一和第二滤波器的传递函数中，以增加信号处理系统的稳定性。

5. 发明授权

US4852175A Hearing aid signal-processing system 失效
标题翻译：助听器信号处理系统
公开(公告)号：US4852175A
公开(公告)日：1989-07-25
申请号：US151850
申请日：1988-02-03
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H04R25/04 , H03G3/20 , H03H17/02 , H04R25/00
CPC分类号： H04R25/356 , H03G3/3089 , H04R2225/43 , H04R25/505
摘要： A signal processing system for hearing aids estimates the absolute quantity of noise in each of a plurality of frequency bands. The audio gains are adjusted so as to minimize the effect of upward spread of masking, wherein noise in lower frequency bands decreases the intelligibility of speech sounds in higher frequency bands. The noise level in each frequency band is estimated by monitoring the amplitude distribution of sound events in that frequency band.
摘要翻译：用于助听器的信号处理系统估计多个频带中的每个频带中的绝对绝对量。调整音频增益以最小化掩蔽向上扩展的影响，其中较低频带中的噪声降低了较高频带中语音的可懂度。通过监测该频带中的声音事件的幅度分布来估计每个频带中的噪声电平。

6. 发明授权

US4468804A Speech enhancement techniques 失效
标题翻译：语音增强技术
公开(公告)号：US4468804A
公开(公告)日：1984-08-28
申请号：US352958
申请日：1982-02-26
申请人： James M. Kates , Julian J. Bussgang
发明人： James M. Kates , Julian J. Bussgang
IPC分类号： G10L21/02 , G10L1/00
CPC分类号： G10L21/02
摘要： A method for processing a voiced speech waveform when the periods and amplitudes thereof may be non-uniform so that the intelligibility thereof is adversely affected. In accordance with such method successive portions of the speech waveform are processed so that each portion has a substantially uniform period and the intelligibility thereof is enhanced. In some instances the processing may be such as to provide in addition substantially uniform peak amplitudes in each processed portion. The voiced speech waveform enhancement technique may further be used in conjunction with methods for processing unvoiced speech waveforms so as to enhance the intelligibility thereof.
摘要翻译：当其周期和幅度可能不均匀以便其可懂度受到不利影响时，处理有声语音波形的方法。根据这种方法，处理语音波形的连续部分，使得每个部分具有基本上均匀的周期，并且其可懂度增强。在一些情况下，处理可以是在每个处理部分中提供基本均匀的峰值振幅。有声语音波形增强技术还可以结合用于处理清音语音波形的方法来使用，以增强其清晰度。

7. 发明授权

US07343022B2 Spectral enhancement using digital frequency warping 有权
标题翻译：使用数字频率翘曲进行光谱增强
公开(公告)号：US07343022B2
公开(公告)日：2008-03-11
申请号：US11226666
申请日：2005-09-13
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H04R25/00
CPC分类号： H04R25/353 , G10H1/125 , G10H2250/115 , G10H2250/235 , H03G7/007 , H04R3/00 , H04R25/356 , H04R25/505 , H04R2225/43
摘要： A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The gain versus frequency function is a set of pure real numbers, so the inverse frequency domain transform gives a set of time-domain filter coefficients. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.
摘要翻译：提供了使用逐采样或块处理的频率扭曲处理系统。这样的系统可以用于例如助听器中以增加语音频谱中的动态范围对比度，从而提高听力容易度和可能的语音清晰度。处理系统由提供频率翘曲的全通滤波器级联组成。从翘曲序列计算功率谱，然后根据听觉分析带的翘曲功率谱计算压缩增益。在翘曲序列中也计算光谱增强增益，允许产生净压缩增强增益增益函数。增益与频率函数是一组纯实数，因此逆频域变换给出了一组时域滤波器系数。语音片段与翘曲时域中的增强滤波器进行卷积，以给出经处理的输出信号。由于频率翘曲系统没有时间混叠，因此减少了处理伪像。

8. 发明授权

US4454609A Speech intelligibility enhancement 失效
标题翻译：语音清晰度增强
公开(公告)号：US4454609A
公开(公告)日：1984-06-12
申请号：US308273
申请日：1981-10-05
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： G10L13/00 , G10L21/02 , G10L1/00 , H04R29/00
CPC分类号： G10L21/0364 , G10L21/0232 , H04R2225/43
摘要： In a communications system, consonant high frequency sounds are enhanced: the greater the high frequency content relative to the low, the more such high frequency content is boosted.
摘要翻译：在通信系统中，辅音高频声音得到增强：相对于低频的高频内容越大，增加的频率越高。

9. 发明授权

US4434482A Tone arm pick up system 失效
标题翻译：调音臂拾取系统
公开(公告)号：US4434482A
公开(公告)日：1984-02-28
申请号：US43353
申请日：1979-05-29
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： G11B3/10
CPC分类号： G11B3/10
摘要： A tone arm system for a high fidelity record player is provided having servo-controlled record warp compensation means. One or more feedback loops are employed to apply a variable force vertically and optionally, laterally, to the tone arm in response to the frequency of stylus vibration. The action of the feedback loop or loops is used to modify the parameters of the tone arm system to provide a predetermined resonance quality factory and a predetermined resonant frequency.
摘要翻译：提供了一种用于高保真记录播放器的音臂系统，具有伺服控制记录扭曲补偿装置。采用一个或多个反馈环来响应于触针振动的频率将可变力垂直地和可选地横向地施加到音调臂。反馈回路或循环的动作用于修改音调臂系统的参数以提供预定的共振质量出厂和预定的共振频率。

10. 发明授权

US4130727A Loudspeaker equalization 失效
标题翻译：扬声器均衡
公开(公告)号：US4130727A
公开(公告)日：1978-12-19
申请号：US811894
申请日：1977-06-29
申请人： James M. Kates
发明人： James M. Kates
IPC分类号： H03G5/02 , H04R3/04
CPC分类号： H03G5/025 , H04R3/04
摘要： Variations in the amplitude-frequency response characteristic of a loudspeaker are compensated by a drive circuit in which an audio input signal is summed with a delayed signal derived from the input signal to provide a drive signal for the loudspeaker. The delayed signal is derived by discrete time filtering and may be formed by storing successively sampled values of the input or drive signal in either digital or analog form, and subsequently recalling the stored values in sequence, or by passing the input or drive signal through a series of all-pass phase-shift filters.
摘要翻译：扬声器的振幅 - 频率响应特性的变化由驱动电路补偿，其中音频输入信号与从输入信号导出的延迟信号相加，以提供用于扬声器的驱动信号。延迟信号是通过离散时间滤波导出的，并且可以通过以数字或模拟形式存储输入或驱动信号的连续采样值，然后依次回忆存储的值，或者通过将输入或驱动信号传递通过系列全通相移滤波器。

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