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    • 1. 发明申请
    • BINAURAL SIGNAL ENHANCEMENT SYSTEM
    • BINAURAL信号增强系统
    • US20080212811A1
    • 2008-09-04
    • US12029292
    • 2008-02-11
    • James M. KATES
    • James M. KATES
    • H04R25/00
    • H04R25/407H04R25/552H04R2225/41
    • 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.
    • 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。
    • 2. 发明申请
    • DYNAMIC RANGE COMPRESSION USING DIGITAL FREQUENCY WARPING
    • 使用数字频率加热的动态范围压缩
    • US20080175422A1
    • 2008-07-24
    • US11866193
    • 2007-10-02
    • James M. KATES
    • James M. KATES
    • H04R25/00
    • H04R25/353G10H1/125G10H2250/115G10H2250/235H03G7/007H04R3/00H04R25/356H04R25/505H04R2225/43
    • A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.
    • 提供动态范围压缩系统,使用逐个采样或块处理系统。 这样的系统可以用于例如助听器中。 使用频率翘曲处理系统的系统由全通滤波器级联组成,全通滤波器的输出提供用于计算滤波器系数的频率分析的输入。 然后在频域中设计压缩滤波器。 使用具有均匀对称性的压缩滤波器可以确保组延迟是恒定的,并且不依赖于任何给定时间的压缩增益。 另外,由于使用全通滤波器,压缩滤波器组延迟更接近于人类听觉延迟。 使用逆向频率变换回翘曲时域来产生与全通延迟线的输出进行卷积的压缩滤波器系数,以给出经处理的输出信号。
    • 3. 发明授权
    • Binaural compression system
    • 双耳压缩系统
    • US07630507B2
    • 2009-12-08
    • US10353187
    • 2003-01-27
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04B1/64H03G7/06H04R25/356H04R25/552
    • 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.
    • 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
    • Binaural signal enhancement system
    • 双耳信号增强系统
    • US07330556B2
    • 2008-02-12
    • US10407305
    • 2003-04-03
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04R25/407H04R25/552H04R2225/41
    • 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.
    • 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。
    • 6. 发明授权
    • Speech enhancement techniques
    • 语音增强技术
    • US4468804A
    • 1984-08-28
    • US352958
    • 1982-02-26
    • James M. KatesJulian J. Bussgang
    • James M. KatesJulian J. Bussgang
    • G10L21/02G10L1/00
    • G10L21/02
    • 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.
    • 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
    • Binaural signal enhancement system
    • 双耳信号增强系统
    • US08036404B2
    • 2011-10-11
    • US12029292
    • 2008-02-11
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04R25/407H04R25/552H04R2225/41
    • 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.
    • 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。
    • 10. 发明授权
    • Dynamic range compression using digital frequency warping
    • 使用数字频率翘曲的动态范围压缩
    • US08014549B2
    • 2011-09-06
    • US11866193
    • 2007-10-02
    • James M. Kates
    • James M. Kates
    • H04R25/00
    • H04R25/353G10H1/125G10H2250/115G10H2250/235H03G7/007H04R3/00H04R25/356H04R25/505H04R2225/43
    • A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.
    • 提供动态范围压缩系统,使用逐个采样或块处理系统。 这样的系统可以用于例如助听器中。 使用频率翘曲处理系统的系统由全通滤波器级联组成,全通滤波器的输出提供用于计算滤波器系数的频率分析的输入。 然后在频域中设计压缩滤波器。 使用具有均匀对称性的压缩滤波器可以确保组延迟是恒定的,并且不依赖于任何给定时间的压缩增益。 另外,由于使用全通滤波器,压缩滤波器组延迟更接近于人类听觉延迟。 使用逆向频率变换回翘曲时域来产生与全通延迟线的输出进行卷积的压缩滤波器系数,以给出经处理的输出信号。