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    • 21. 发明申请
    • REAL-TIME TECHNIQUE FOR REDUCING THE SETTLING TIME OF A HIGH PASS FILTER
    • 实时技术降低高通滤波器的设定时间
    • WO00035087A1
    • 2000-06-15
    • PCT/US1999/027707
    • 1999-11-22
    • A61B5/022H03H17/02
    • H03H17/0219
    • A technique for resetting an HPF for improving its settling time in response to a sharp transition in its input signal, by rapid input and clocking through the HPF of new data samples after occurrence of the sharp transition. The new data samples comprise an artifical ramp representative of the expected average slope of the input signal that should be maintained after occurence of the sharp transition. The artificial ramp can be generated using "a priori" knowledge of the expected average slope of the input signal, or alternatively, can be generated using a direct measurement of the slope of the input signal at the time the HPF is to be reset. In the event that both the direct slope measurement and "a priori" slope knowledge are available at the time the HPF is to be reset, and they differ from each other, it is desirable to generate an artificial ramp whose slope provides a smoothed transition from the latter slope measurement to the former slope.
    • 一种技术,用于通过快速输入和通过快速转换发生后的新数据样本的HPF来重新设置HPF,以改善其建立时间,以响应其输入信号的急剧转变。 新的数据样本包括代表在发生急剧转变之后应该保持的输入信号的预期平均斜率的人造斜坡。 可以使用对输入信号的预期平均斜率的“先验”知识来生成人造斜坡,或者可以使用在HPF被复位时直接测量输入信号的斜率来生成。 在HPF要复位时,直接斜率测量和“先验”斜率知识都可用,并且它们彼此不同的情况下,希望产生一个人造斜坡,其斜坡提供平滑的过渡 后者斜率测量到前一坡。
    • 23. 发明申请
    • MAGNITUDE COMPENSATION TECHNIQUE FOR PROCESSING SINGLE-BIT WIDE DATA
    • 单位宽数据处理的幅度补偿技术
    • WO2017180172A1
    • 2017-10-19
    • PCT/US2016/038545
    • 2016-06-21
    • CIRRUS LOGIC INTERNATIONAL SEMICONDUCTOR LTD.HESTER, Dylan, Alexander
    • HESTER, Dylan, Alexander
    • H03H17/02H03H17/06H03M1/66
    • H03M1/0626H03H17/0219H03M1/66
    • Droop caused by a filter may be compensated by applying a pre-filter to the audio signal that cancels out, at least in part, the droop caused by the filter. The pre-filter may implement magnitude compensation that causes an approximately flat passband response when the pre-filtered signal is passed through the filter. The pre-filter may be applied to one-bit wide data streams, such as high-fidelity direct stream digital (DSD) audio data or other one-bit wide data such as pulse-density modulation (PDM) encoded data. The pre-filtering and filtering may be implemented in components of an audio processor, such as in a digital-to-analog converter (DAC). The pre-filtering may include upsampling the one-bit wide data to form symbols and substituting an eighth bit of the symbol with an inverted version of an earlier-received bit.
    • 由滤波器引起的下垂可以通过对音频信号应用预滤波器来补偿,所述预滤波器至少部分抵消由滤波器引起的下垂。 预滤波器可以实施幅度补偿,当预滤波信号通过滤波器时,该幅度补偿导致近似平坦的通带响应。 预滤波器可以应用于诸如高保真直接流数字(DSD)音频数据的一位宽数据流或诸如脉冲密度调制(PDM)编码数据的其他一位宽数据。 预滤波和滤波可以在音频处理器的组件中实现,例如在数模转换器(DAC)中。 预滤波可以包括对一位宽数据进行上采样以形成符号,并用先前接收的比特的反转版本替换符号的第八位。
    • 27. 发明申请
    • SAMPLING RATE CONVERTER DATA FLOW CONTROL MECHANISM
    • 采样率转换器数据流量控制机制
    • WO2011085947A1
    • 2011-07-21
    • PCT/EP2011/000034
    • 2011-01-07
    • ST-ERICSSON SATUDOSE, Andrei
    • TUDOSE, Andrei
    • H03H17/06H04L7/00
    • H04L7/0029H03H17/0219H03H17/0621H03H17/0657
    • Sampling rate converter for performing a rate conversion of an incoming stream of data, clocked at a first frequency, and output at a second frequency; said converter comprising: a circuit (2) for up-sampling the incoming data stream; a filter (3) for filtering the up-sampled incoming data stream; an interpolation filter (4) for interpolating the filtered up-sampled incoming stream of data; a FIFO (5) for storing the interpolated filtered up-sampled incoming stream of data, said storing being performed at said first frequency and the reading of said FIFO being performed at said second frequency so as to output the stored data at a second corresponding rate; characterized in that it further comprises: a control block (10) comprising a numerically controlled oscillator (NCO) used for generating said first frequency, the control of said NCO being based on information representative of the status of said FIFO (5) and also information representative of the modulation of the data and ratio of sampling rate change.
    • 采样率转换器,用于执行以第一频率计时并以第二频率输出的输入数据流的速率转换; 所述转换器包括:用于对输入数据流进行上采样的电路(2); 滤波器(3),用于过滤上采样的输入数据流; 内插滤波器(4),用于内插经滤波的上采样的数据流; 一个FIFO(5),用于存储内插滤波的上采样输入数据流,所述存储以所述第一频率执行,并且所述FIFO的读取以所述第二频率执行,以便以第二对应速率输出所存储的数据 ; 其特征在于,还包括:包括用于产生所述第一频率的数控振荡器(NCO)的控制块(10),所述NCO的控制是基于表示所述FIFO(5)的状态的信息,以及信息 代表数据的调制和采样率的变化率。
    • 28. 发明申请
    • 音質調整装置及びこれに用いるフィルタ装置、音質調整方法、フィルタの設計方法
    • 声音质量调节装置及其使用的滤波器,声音质量调节方法和滤波器设计方法
    • WO2003023961A1
    • 2003-03-20
    • PCT/JP2002/009047
    • 2002-09-05
    • 酒井 康江小柳 裕喜生
    • 小柳 裕喜生
    • H03H17/06
    • H03G3/002H03H17/0219H03H17/06
    • A filter circuit is composed of FIR filters, and the filter factor is of a symmetrical type, thereby realizing a linear phase characteristic and preventing phase distortion even if a desired frequency band is emphasized. The first filter factors of a low-pass filter are determined so that the first filter factors may be the terms of a sequence in which the sum of the terms is not zero, the sign of the sum of every other terms is the same as that of the other every other terms, and the sum of the every other terms is equal to that of the other every other terms. Thus, by only changing a part of the signs of the filter factors, second filter factors of a high-pass filter can be simply determined so that they may be the terms of a sequence in which the sum of the terms is zero, the sign of the sum of every other terms is opposite to that of the other every other terms, and the absolute value of the sum of the every other terms is equal to that of the other every other terms.
    • 滤波器电路由FIR滤波器构成,滤波器因子是对称型,从而实现线性相位特性,并且即使强调期望的频带也能防止相位失真。 确定低通滤波器的第一滤波器因子,使得第一滤波器因子可以是术语的和不为零的序列的项,每个其他项的和的符号与 的其他条款,其他条款的总和等于其他条款的总和。 因此,通过仅改变滤波器因子的一部分符号,可以简单地确定高通滤波器的第二滤波器因子,使得它们可以是项的和为零的序列的项,符号 每个其他条款的总和与其他条款的总和相反,每个其他条款的总和的绝对值等于其他每个条款的总和的绝对值。
    • 30. 发明申请
    • APPARATUS FOR AND METHODS OF DIGITAL PROCESSING
    • 数字处理的方法和方法
    • WO1980000393A1
    • 1980-03-06
    • PCT/GB1979000129
    • 1979-08-02
    • UNIVERSITY OF SURREYMATLEY WWILLIAMSON SBYWATER R
    • UNIVERSITY OF SURREY
    • H03H17/02
    • H03H17/026H03H17/0219
    • A digital filter that removes from a digitised analogue signal those frequency components that would give rise to aliasing in a subsequent digital filtering operation, and does so without phase distortion. In a first stage (S1) of the filter successive overlapping groups of digitised sampled values are averaged, using a random access memory (2), an adder unit (3) and a parallel shift register (9). In a second stage (S2) successive non-overlapping groups of the averaged sampled values are accumulated (12) and averaged in a further parallel shift register (13). The second stage output may be converted to an analogue signal or supplied to a main digital filter (MDF), and the filter is particularly suited to microprocessor applications.
    • 数字滤波器,从数字化模拟信号中去除那些会在随后的数字滤波操作中产生混叠的频率分量,并且没有相位失真。 在滤波器的第一级(S1)中,使用随机存取存储器(2),加法器单元(3)和并行移位寄存器(9)对数字化采样值的连续重叠组进行平均。 在第二级(S2)中,平均采样值的连续非重叠组被累加(12)并在另一个并行移位寄存器(13)中进行平均。 第二级输出可以转换为模拟信号或提供给主数字滤波器(MDF),并且滤波器特别适用于微处理器应用。