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    • 1. 发明授权
    • Nonrecursive digital filter and method for calculating the coefficients of the filter
    • 非递归数字滤波器和计算滤波器系数的方法
    • US06446103B2
    • 2002-09-03
    • US09752923
    • 2001-01-02
    • Peter Schöllhorn
    • Peter Schöllhorn
    • G06F1710
    • H03H17/0621
    • The nonrecursive digital filter has variable coefficients for the purpose of shifting the frequency response from one characteristic frequency fc1 to another characteristic frequency fc2. The method enables calculating the coefficients of the nonrecursive digital filter for shifting the frequency response of the filter from one characteristic frequency fc1 to another characteristic frequency fc2. The coefficients of a prototype of the filter are prescribed or calculated for a given sampling frequency, the continuous time impulse response associated with this prototype is calculated and the impulse response is sampled at the sampling frequency fa2=fa1* fc1/fc2. The new coefficients are thereby produced. The filter can then be operated with the new coefficients at the old sampling frequency. As an alternative, the new coefficients can be calculated from the coefficients of the prototype of the filter by interpolating the values of the continuous time impulse response at an interval of 1/fa2.
    • 非递归数字滤波器具有可变系数,用于将频率响应从一个特征频率fc1移位到另一个特征频率fc2。 该方法能够计算用于将滤波器的频率响应从一个特征频率fc1移位到另一个特征频率fc2的非递归数字滤波器的系数。 对于给定的采样频率规定或计算滤波器原型的系数,计算与该原型相关联的连续时间脉冲响应,并以采样频率fa2 = fa1 * fc1 / fc2对脉冲响应进行采样。 由此产生新的系数。 然后可以用旧采样频率的新系数来操作滤波器。 作为替代,可以通过以1 / fa2的间隔内插连续时间脉冲响应的值,从滤波器的原型的系数来计算新系数。
    • 2. 发明授权
    • Analogue frontend
    • 模拟前端
    • US06522272B2
    • 2003-02-18
    • US09780936
    • 2001-02-09
    • Andreas MenkhoffPeter Schöllhorn
    • Andreas MenkhoffPeter Schöllhorn
    • H03M184
    • H04B1/0007H04B1/18H04B1/30
    • An analogue frontend, such as is used, for example, in receivers for communications equipment, comprises a plurality of analogue components (1-3) which are connected upstream of an analogue/digital converter (4). In order to control the drive level of the individual analogue components (1-3) precisely, the input of the analogue/digital converter (4) can optionally be connected via a switch-over device (8) to the output of any one of the analogue components (1-3) and a drive level control signal for the respective analogue component (1-3) can be generated by a digital control device (5) by evaluating the sampled value which is then supplied by the analogue/digital converter (4).
    • 诸如在用于通信设备的接收机中使用的模拟前端包括连接在模拟/数字转换器(4)的上游的多个模拟组件(1-3)。 为了精确地控制各个模拟部件(1-3)的驱动电平,模拟/数字转换器(4)的输入可以可选地通过切换装置(8)连接到任何一个的输出 数字控制装置(5)可以通过对采样值进行评估,然后由模拟/数字转换器(1-3)提供模拟组件(1-3)和相应的模拟组件(1-3)的驱动电平控制信号 (4)。
    • 4. 发明授权
    • Circuit for demodulating two-dimensional data symbols transmitted by a carrier-based data transmission
    • 用于解调通过基于载波的数据传输发送的二维数据符号的电路
    • US06424683B1
    • 2002-07-23
    • US09625217
    • 2000-07-24
    • Peter Schöllhorn
    • Peter Schöllhorn
    • H03D322
    • H04L27/38
    • A circuit for demodulating two-dimensional data symbols transmitted by carrier-based data transmission. The circuit has a tuner and a quadrature demodulator. The tuner has a first intermediate frequency stage for converting a carrier frequency of an input signal containing the two-dimensional data symbols to a first intermediate frequency f1, a surface wave filter, and a second intermediate frequency stage connected downstream of the surface wave filter for converting the carrier frequency of the input signal to a second intermediate frequency f2. The second intermediate frequency f2 is lower than the first intermediate frequency f1. The demodulator converts the input signal at the second carrier intermediate frequency f2 to a baseband and splits it into in-phase and quadrature components. The quadrature demodulator contains an A/D converter with a prescribed sampling frequency fa, and an anti-alias filter is connected upstream of the A/D converter in order to remove interfering spectral components from the useful signal. The following relationship between the first and second intermediate frequencies f1, f2 and the sampling frequency fa are defined by the equations: f2=f1/5 and fa=4·f2 or fa=4/5·f1.
    • 用于解调通过基于载波的数据传输发送的二维数据符号的电路。 该电路具有调谐器和正交解调器。 调谐器具有用于将包含二维数据符号的输入信号的载波频率转换为连接在表面波滤波器下游的第一中频f1,表面波滤波器和第二中频级的第一中频级, 将输入信号的载波频率转换为第二中频f2。 第二中频f2低于第一中频f1。 解调器将第二载波中频f2处的输入信号转换为基带并将其分为同相和正交分量。 正交解调器包含具有规定采样频率fa的A / D转换器,并且在A / D转换器的上游连接抗混叠滤波器,以从有用信号中去除干扰频谱分量。 第一和第二中间频率f1,f2和采样频率fa之间的以下关系由等式定义:
    • 5. 再颁专利
    • Method for selective filtering
    • 选择性过滤方法
    • USRE39283E1
    • 2006-09-12
    • US10322081
    • 2002-12-17
    • Peter Schöllhorn
    • Peter Schöllhorn
    • G06F17/10
    • H04L7/0029
    • A method for digital clock recovery and selective filtering includes prescribing or calculating first coefficients of a prototype of a selective filter at a characteristic frequency fc for a given sampling frequency fa. Second coefficients of a selective filter are calculated at the characteristic frequency fc2 from the first coefficients at points tk=Δt+k*d, where k is an integer greater than or equal to 0, by interpolating values of a continuous-time impulse response at points tk, where d = fc2 * fa fc * fa 1 The selective filter is operated with the second coefficients at fa1.
    • 一种用于数字时钟恢复和选择性滤波的方法包括在给定采样频率fa的特征频率fc处处理或计算选择滤波器的原型的第一系数。 选择滤波器的第二系数是在特征频率fc 2处,通过内插的方式,从点t k = Deltat + k * d处的第一系数计算,其中k是大于或等于0的整数 在点t k处的连续时间脉冲响应的值,其中 < MI> d = fc2 选择性滤波器以第一系数在fa 1> 1操作。
        • 6. 发明授权
        • Method for selective filtering
        • 选择性过滤方法
        • US06438567B2
        • 2002-08-20
        • US09752922
        • 2001-01-02
        • Peter Schöllhorn
        • Peter Schöllhorn
        • G06F1710
        • H04L7/0029
        • A method for digital clock recovery and selective filtering includes prescribing or calculating first coefficients of a prototype of a selective filter at a characteristic frequency fc for a given sampling frequency fa. Second coefficients of a selective filter are calculated at the characteristic frequency fc2 from the first coefficients at points tk=&Dgr;t+k*d, where k is an integer greater than or equal to 0, by interpolating values of a continuous-time impulse response at points tk, where d = fc2 * fa fc * fa 1 . The selective filter is operated with the second coefficients at fa1.
        • 一种用于数字时钟恢复和选择性滤波的方法包括在给定采样频率fa的特征频率fc处处理或计算选择滤波器的原型的第一系数。 选择滤波器的第二系数通过将连续时间脉冲响应的值内插到位于tk = DELTAt + k * d处的第一系数的特征频率fc2,其中k是大于或等于0的整数 点tk,其中选择性滤波器以fa1的第二系数运行。