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    • 2. 发明授权
    • Method for determining unbiased signal amplitude estimates after cepstral variance modification
    • 倒谱方差修正后确定无偏信号幅度估计的方法
    • US08208666B2
    • 2012-06-26
    • US12684147
    • 2010-01-08
    • Timo GerkmannRainer Martin
    • Timo GerkmannRainer Martin
    • H04R25/00
    • G10L21/0208G10L25/24
    • A method for determining unbiased signal amplitude estimates () after cepstral variance modification of a discrete time domain signal (s(t)), wherein the cepstrally-modified spectral amplitudes () of the discrete time domain signal (s(t)) are χ-distributed with 2{tilde over (μ)} degrees of freedom. A bias reduction factor (r) is determined using the equation r 2 = μ μ ~ ⁢ ⅇ ψ ⁡ ( μ ~ ) - ψ ⁡ ( μ ) , where 2μ are the degrees of freedom of the χ-distributed spectral amplitudes of the discrete time domain signal (s(t)) and ψ ⁡ ( x ) = - 0.5772 - ∑ n = 0 ∞ ⁢ ( 1 x + n - 1 1 + n ) ; then the unbiased signal amplitude estimates () are determined by multiplying the cepstrally-modified spectral amplitudes () with the bias reduction factor (r) according to the equation =r. A method for speech enhancement and a hearing aid use the method for determining unbiased signal amplitude estimates () in order to offer the advantage of spectral modification, such as smoothing, of spectral quantities without affecting their signal power.
    • 一种用于确定离散时域信号(s(t))的倒谱方差修正后的无偏信号幅度估计()的方法,其中离散时域信号(s(t))的经频谱修正的频谱幅度(χ)为χ 分布为2(波形)超过(μ)}自由度。 使用等式r 2 =μμ〜ψψ(μ〜)-ψ⁡(μ)确定偏置约简因子(r),其中2μ是离散时间的χ分布频谱幅度的自由度 域信号(s(t))和ψ⁡(x)= - 0.5772 - Σn = 0∞(1 x + n - 1 1 + n) 那么无偏信号幅度估计()通过将倒频修正的频谱幅度()与偏置简化因子(r)相乘来确定,方程式= r。 用于语音增强和助听器的方法使用用于确定无偏信号幅度估计()的方法,以便提供频谱修改的优点,例如频谱量化的平滑,而不影响其信号功率。
    • 4. 发明申请
    • Pressure sensor
    • 压力传感器
    • US20070028696A1
    • 2007-02-08
    • US10566364
    • 2004-07-16
    • Ulfert DrewesKarl FloegelFrank HegnerThomas UehlinRainer Martin
    • Ulfert DrewesKarl FloegelFrank HegnerThomas UehlinRainer Martin
    • G01L7/00
    • G01L19/003
    • A pressure sensor including: A pressure measuring cell having an end face loadable with the medium; a housing having a media opening and a ring-shaped, axial bearing surface, which surrounds the media opening; a clamping apparatus; and a ring-shaped sealing arrangement. The pressure measuring cell is positioned in the housing and the sealing arrangement is positioned between the bearing surface and the end face, and the sealing arrangement, as well as the pressure measuring cell, are axially clamped between the bearing surface and the clamping apparatus. The sealing arrangement includes, a decoupling ring, as well as a first and a second, ring-shaped sealing element. The first sealing element lies against the end face, the second sealing element lies against the bearing surface, and the decoupling ring is axially clamped between the first and the second sealing elements.
    • 一种压力传感器,包括:压力测量单元,其具有可与介质一起装载的端面; 壳体,其具有介质开口和围绕介质开口的环形的轴向支承表面; 夹紧装置; 和环形密封装置。 压力测量单元位于壳体中,并且密封装置位于支承表面和端面之间,并且密封装置以及压力测量单元被轴向地夹紧在支承表面和夹紧装置之间。 密封装置包括脱钩环以及第一和第二环形密封元件。 第一密封元件靠在端面上,第二密封元件位于支承表面上,并且解耦环被轴向夹紧在第一和第二密封元件之间。
    • 6. 发明授权
    • Apparatus for improving disturbed speech signals
    • 用于改善受干扰的语音信号的装置
    • US5699480A
    • 1997-12-16
    • US673151
    • 1996-07-01
    • Rainer Martin
    • Rainer Martin
    • G10L21/0208G10L21/0216H03H17/00H03H21/00G10L3/02
    • G10L21/0208G10L2021/02165
    • An apparatus for the transmission of speech signals reduces the effect of disturbances on the transmission quality. Two microphones are provided for picking up speech signals. Signal processing ensues in three partial frequency bands. The microphone signals are high-pass filtered in a lower frequency band, in the middle frequency band the signal is weighted with a scalar factor, so that this frequency band is damped during the speech pauses, and in the upper frequency band, an adaptive filter is used. At the beginning of the processing, a treble enhancement of the signals ensues, which is canceled by an inverse filter before the output of the improved signal.
    • 用于传输语音信号的装置减少了干扰对传输质量的影响。 提供两个麦克风用于拾取语音信号。 在三个部分频带中进行信号处理。 麦克风信号在较低频带中被高通滤波,在中频段,信号用标量因子加权,使得该频带在语音暂停期间被衰减,而在较高频带中,自适应滤波器 用来。 在处理开始时,信号的高音增强,在改善的信号的输出之前由逆滤波器抵消。