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    • 14. 发明授权
    • Filter circuit with reduced number of delay elements and adders
    • 具有减少数量的延迟元件和加法器的滤波电路
    • US5886914A
    • 1999-03-23
    • US892968
    • 1997-07-15
    • Satoshi SugawaShiro Hosotani
    • Satoshi SugawaShiro Hosotani
    • G06F17/15H03H17/02G06F17/10
    • H03H17/0238G06F17/15
    • An output from an adder 10.sub.1, i.e., an output of a first bit plane is inputted to an adder 10.sub.6 through delay elements 2.sub.2 and 3.sub.0 and a multiplier 100.sub.0. On the other hand, input data X are inputted to multipliers C.sub.2.sup.1 to C.sub.0.sup.1 through a delay element 1.sub.0 and multiplied by the respective multipliers to obtain partial products. An adder 10.sub.2 receives a partial product by the multiplier C.sub.2.sup.1 through a delay element 2.sub.3 and a partial product by the multiplier C.sub.1.sup.1. An adder 10.sub.3 receives an output from the adder 10.sub.2 through the delay element 2.sub.3 and on the other hand a partial product by the multiplier C.sub.0.sup.1. An output from the adder 10.sub.3, i.e., an output of a second bit plane is inputted to the adder 10.sub.6 through a delay element 2.sub.5. The adder 10.sub.6 performs addition of the output from the adder 10.sub.3, i.e., the output of the second bit plane and the output from the adder 10.sub.1, i.e., the output of the first bit plane, to output the addition result. This structure allows reduction in the number of delay elements and adders, to achieve a filter circuit downsized in circuit scale.
    • 来自加法器101的输出即第一位平面的输出通过延迟元件22和30以及乘法器1000输入到加法器106.另一方面,输入数据X通过一个输入数据X输入到乘法器C21至C01 延迟元件10并乘以相应的乘法器以获得部分乘积。 加法器102通过延迟元件23由乘法器C21接收部分积,乘法器C11接收部分乘积。 加法器103通过延迟元件23接收来自加法器102的输出,另一方面由乘法器C01接收部分乘积。 来自加法器103的输出即第二位平面的输出通过延迟元件25输入到加法器106.加法器106执行加法器103的输出,即第二位平面的输出 和加法器101的输出,即第一位平面的输出,以输出相加结果。 这种结构允许减少延迟元件和加法器的数量,以实现电路规模缩小的滤波器电路。
    • 17. 发明申请
    • ADAPTIVE FILTER SYSTEM HAVING MIXED FIXED POINT OR FLOATING POINT AND BLOCK SCALE FLOATING POINT OPERATORS
    • 具有混合固定点或浮点的自适应滤波系统和块尺度浮点运算符
    • WO99007071A1
    • 1999-02-11
    • PCT/US1998/013838
    • 1998-07-02
    • H03H17/02H03H17/06H03H21/00H04B3/23
    • H03H17/0238H03H17/0227H03H21/0012
    • An adaptive filter is provided which has fixed point or floating point data stored in a data RAM (74) and block scale floating point coefficients stored in a coefficient RAM (84). The data and coefficients are utilized in a filter algorithm which utilizes a multiplier and an accumulator to provide a convolution result. Coefficients are uptdated by adding the multiplied result of the data RAM value and the error value to the old value of the coefficient. This is done for all the coefficient values in the coefficient RAM. The error value indicates the difference between the filter output and the sample near-end signal that is the echo. These new coefficients are examined and if any have a value above or all have a value below a predetermined threshold, then the mantissas of all the coefficents are shifted and the exponent adjusted in the next filter cycle.
    • 提供了一种自适应滤波器,其具有存储在数据RAM(74)中的固定点或浮点数据以及存储在系数RAM(84)中的块比例浮点系数。 数据和系数被用在利用乘法器和累加器来提供卷积结果的滤波器算法中。 通过将数据RAM值的相乘结果和误差值相加到系数的旧值来升高系数。 这是对系数RAM中的所有系数值进行的。 误差值表示滤波器输出与作为回波的样本近端信号之间的差异。 检查这些新系数,并且如果有的值高于或全部具有低于预定阈值的值,则所有系数的尾数被移动,并且在下一个滤波周期中调整指数。
    • 20. 发明专利
    • Signal processor, television, signal processing method, program and recording medium
    • 信号处理器,电视,信号处理方法,程序和记录介质
    • JP2012039232A
    • 2012-02-23
    • JP2010175389
    • 2010-08-04
    • Sharp Corpシャープ株式会社
    • MURAHASHI YOSHIMITSU
    • H03H17/02G10L21/02G10L21/04H04R3/04
    • H03H17/0461H03H17/0238H03H17/0263H03H17/0286H03H2017/0295H03H2017/0298H03H2017/0477H04R3/00H04R2499/15
    • PROBLEM TO BE SOLVED: To provide a signal processor for making the amount of noise generated in signal processing smaller than before even when it is provided as a signal processor including an inexpensive fixed point operation DSP.SOLUTION: A sound processing section 112 includes: an equalizer section 1121 for making n pieces of biquadratic filters 1121-1 to 1121-n having independent filter characteristics successively act on sound signals; a noise level estimation section 1124 for estimating the size of noise to be outputted from the equalizer section 1121 by making the n pieces of biquadratic filters act in the order to the sound signals for permutation which can be constituted of the n pieces of biquadratic filters; and a filter coefficient setting section 1123 for practically changing the order of making the n pieces of biquadratic filters act on input signals by changing the filter coefficients of the respective biquadratic filters so as to reduce the noise outputted from the equalizer section 1121 on the basis of the estimated size of the noise.
    • 要解决的问题:提供一种信号处理器,用于使信号处理中产生的噪声量比以前更小,即使当其被提供为包括便宜的定点运算DSP的信号处理器时。 解决方案:声音处理部分112包括:均衡器部分1121,用于使n个具有独立滤波器特性的双二次滤波器1121-1至1121-n依次作用在声音信号上; 噪声电平估计部分1124,用于通过使得n个双二次滤波器按照可以由n个双二次滤波器构成的置换声音信号的顺序来估计从均衡器部分1121输出的噪声的大小; 以及滤波器系数设定部分1123,用于通过改变各个二次滤波器的滤波器系数来实际改变使n个双二进制滤波器的顺序对输入信号起作用,以便基于均衡器部分1121输出的噪声 估计的噪音大小。 版权所有(C)2012,JPO&INPIT