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    • 1. 发明授权
    • Multiple description coding communication system
    • 多描述编码通信系统
    • US06823018B1
    • 2004-11-23
    • US09511367
    • 2000-02-23
    • Hamid JafarkhaniMichael T. OrchardAmy R. ReibmanYao Wang
    • Hamid JafarkhaniMichael T. OrchardAmy R. ReibmanYao Wang
    • H04B1406
    • H04N19/39
    • A method and apparatus reliably encode and decode information over a communication system. The method includes transforming two coefficients into two pairs of random variables, one random variable in each pair having substantially equal energy as one random variable in the other pair. The method further includes quantizing each of the pairs of random variables and entropy coding each quantized random variable separately creating an encoded bitstreams. The encoded bitstreams are received by a decoder which first determines which channels of the communication system are working. The encoded bitstream is entropy decoded, inversed quantized and inversed transformed. An inverse transform performs three different transformations depending upon which channels are working, i.e., whether the first, second or both channels are working.
    • 一种方法和装置通过通信系统可靠地编码和解码信息。 该方法包括将两个系数变换成两对随机变量,每对中的一个随机变量具有基本相等的能量作为另一对中的一个随机变量。 该方法还包括量化每对随机变量,并对每个量化的随机变量进行熵编码,分别创建编码比特流。 编码的比特流由解码器接收,该解码器首先确定通信系统的哪些信道正在工作。 编码比特流被熵解码,反相量化和反转换。 逆变换根据哪个通道工作,即第一,第二或两个通道是否工作,执行三种不同的变换。
    • 4. 发明授权
    • Voice recording and playback mode using the G.726 half-rate within the personal handy phone system
    • 语音录音和播放模式使用个人手机系统内的G.726半速率
    • US06574281B1
    • 2003-06-03
    • US09753824
    • 2001-01-02
    • Satoshi YoshidaPatrick FeyfantPhilippe GaglioneDenis ArchambaudVarenka MartinLaurent WinckelRita LagomarsinoOliver Weigelt
    • Satoshi YoshidaPatrick FeyfantPhilippe GaglioneDenis ArchambaudVarenka MartinLaurent WinckelRita LagomarsinoOliver Weigelt
    • H04B1406
    • H04M1/725H04B14/068H04M1/6505
    • Voice recording and playback mode using the G.726 half-rate within the personal handy phone system (PHS). When a portable station within the PHS operates as a voice recorder (e.g., functioning as an answering machine), a cost effective system in accordance with the present invention is adapted to compress and store received voice/sound signals in order to increase the usage of limited memory resources provided within the portable station. The present invention also enables previously compressed and stored voice/sound signals to be decompressed and played back in various portable station playback modes. Specifically, the portable station receives a voice/sound signal in a full rate (e.g., 32 kilobits-per-second) 4-bit adaptive differential pulse code modulation (ADPCM) data format in compliance with the International Telecommunication Union (ITU) recommendation G.726. The present invention compresses this received voice/sound signal to a half rate (16 kilobit-per-second) 2-bit ADPCM data format in compliance with the ITU recommendation G.726 in order to increase the usage of the limited memory resources provided within the portable station. During a playback mode of the portable station, the present invention decompresses the previously compressed and stored voice/sound signal to facilitate its playback.
    • 录音和播放模式使用个人手机系统(PHS)内的G.726半速率。 当PHS内的便携式站作为语音记录器(例如,用作应答机)时,根据本发明的成本有效的系统适于压缩和存储接收到的语音/声音信号,以便增加使用 在便携式站内提供的有限的存储器资源。 本发明还使得先前压缩和存储的语音/声音信号能够以各种便携式电台播放模式被解压缩和重放。 具体来说,便携式电台以符合国际电信联盟(ITU)推荐G的全速率(例如,32千比特每秒)4位自适应差分脉码调制(ADPCM)数据格式接收语音/声音信号 .726。 本发明根据ITU推荐G.726将该接收到的语音/声音信号压缩到半速率(16千比特每秒)2位ADPCM数据格式,以便增加在内部提供的有限存储器资源的使用 便携式电台。 在便携式终端的回放模式期间,本发明解压缩先前压缩和存储的语音/声音信号以便于其播放。
    • 6. 发明授权
    • Systems, methods and computer program products for filtering glitches from measured values in a sequence of code points
    • 用于从代码点序列中的测量值过滤毛刺的系统,方法和计算机程序产品
    • US06823017B1
    • 2004-11-23
    • US09432023
    • 1999-10-29
    • Gordon Taylor DavisFredy D. NeeserMalcolm Scott Ware
    • Gordon Taylor DavisFredy D. NeeserMalcolm Scott Ware
    • H04B1406
    • H04B14/046
    • Glitch filters, methods, and computer program products that utilize the generally monotonically increasing characteristics of the expected levels of code points to detect and remove noise spikes by replacing values in the code point sequence with new values based on the code points around a suspect value are provided. Measured values associated with two code points in the sequence of code points which are immediately higher in the sequence of code points than a code point of interest are evaluated so as to select a larger value of the two code points in the sequence as a first reference value. The first reference value is compared with a measured value associated with a code point in the sequence of code points immediately lower than the code point of interest to determine if the first reference value is smaller than the measured value associated with the code point in the sequence of code points immediately lower than the code point of interest. The smaller of the first reference value and the measured value associated with a code point in the sequence of code points immediately lower than the code point of interest is then selected so as to provide a first replacement value. The measured value associated with the code point of interest is then replaced with the first replacement value if the first reference value is smaller than the measured value associated with the code point of interest.
    • 毛刺滤波器,方法和计算机程序产品利用通常单调增加的代码点级别的特征来检测和消除噪声尖峰,通过使用基于可疑值周围的代码点的新值替换代码点序列中的值, 提供。 评估与编码点序列中的代码点比代码点高的代码点序列中的两个代码点相关联的测量值,以选择序列中的两个代码点的较大值作为第一参考 值。 将第一参考值与与紧接着感兴趣码点的代码点序列中的代码点相关联的测量值进行比较,以确定第一参考值是否小于与序列中的代码点相关联的测量值 的代码点立即低于感兴趣的代码点。 然后选择第一参考值和与紧接着感兴趣的代码点的代码点序列中的代码点相关联的测量值中的较小者,以提供第一替换值。 如果第一参考值小于与感兴趣的代码点相关联的测量值,则与感兴趣的代码点相关联的测量值被替换为第一替换值。
    • 8. 发明授权
    • Single bit Sigma Delta filter with input gain
    • 具有输入增益的单位ΣΣ滤波器
    • US06408031B1
    • 2002-06-18
    • US09428086
    • 1999-10-27
    • Paul David Hendricks
    • Paul David Hendricks
    • H04B1406
    • H03H17/0411
    • A digital system for filtering a single bit input signal according to the transfer function H(z), wherein H(z) has a gain G, a pole at location b0, and a zero at location a0. The digital system filters the single bit input signal without using computationally expensive multibit multiplication. The digital system achieves these advantages with a digital circuit having a first gain stage generating a gain corrected signal, a delay element generating a delayed gain corrected signal, a feed-forward stage generating a feed-forward signal, and a summer for generating an output signal based upon the sum of the gain corrected signal, the delayed gain corrected signal and the feed-forward signal.
    • 一种用于根据传递函数H(z)对单个位输入信号进行滤波的数字系统,其中H(z)具有增益G,位置b0处的极点和位置a0处的零点。 数字系统对单位输入信号进行滤波,而不用计算昂贵的多位乘法。 数字系统通过具有产生增益校正信号的第一增益级的数字电路,产生延迟增益校正信号的延迟元件,产生前馈信号的前馈级和用于产生输出的加法器来实现这些优点 基于增益校正信号,延迟增益校正信号和前馈信号之和的信号。
    • 9. 发明授权
    • One bit digital quadrature vector modulator
    • 一位数字正交矢量调制器
    • US06339621B1
    • 2002-01-15
    • US09135243
    • 1998-08-17
    • Christian CojocaruTheodore VarelasMark CloutierLuc Lussier
    • Christian CojocaruTheodore VarelasMark CloutierLuc Lussier
    • H04B1406
    • H04L27/362H03F2200/331
    • A One Bit Digital Quadrature Vector Modulator (DQVM) and a method of generating single sideband output signals are useful for a wide range of radio frequency, signal processing and wireless applications. The DQVM simplifies the necessary digital multiplication by using noise shaped one bit versions of both the baseband IB and QB signals to be modulated and the ILO and QLO modulating signals. The one bit DQVM enables a much faster digital implementation of the digital quadrature vector modulation function than can be achieved with conventional multi-bit digital techniques. In addition the single sideband upconversion of the DQVM achieves high suppression of the unwanted sideband by applying an offset to one of the low speed input samples. Digital vector modulators are an improvement over conventional analog vector modulators as they are not subject to the amplitude and phase matching problems inherent in analog vector modulators.
    • 一个位数字正交矢量调制器(DQVM)和一种产生单边带输出信号的方法可用于广泛的射频,信号处理和无线应用。 DQVM通过使用要调制的基带IB和QB信号的噪声形状的一位版本以及ILO和QLO调制信号来简化必要的数字乘法。 与使用传统的多位数字技术相比,该位DQVM能够实现数字正交矢量调制功能的更快速的数字实现。 此外,DQVM的单边带向上转换通过对低速输入采样之一施加偏移来实现对不想要的边带的高抑制。 数字矢量调制器是对传统模拟矢量调制器的改进,因为它们不受模拟矢量调制器固有的幅度和相位匹配问题的影响。
    • 10. 发明授权
    • Reduced DC transients in a sigma delta filter
    • 减少西格玛滤波器中的直流瞬变
    • US06823019B1
    • 2004-11-23
    • US09363005
    • 1999-07-30
    • Paul D. HendricksDonald R. LaturellLane A. Smith
    • Paul D. HendricksDonald R. LaturellLane A. Smith
    • H04B1406
    • H03H17/04
    • DC transients are removed from a digital filter such as a sigma delta filter (in particular from a sigma delta high pass filter) from the outset by presetting an input summing node to a sigma delta modulator. While the input summing node may be preset using any appropriate input, in a disclosed embodiment, a sigma delta high pass filter is preset by switching a partial feedback term between an input containing the non-zero preset value and the normal input comprising the output from the input summing node. The preset value is chosen based on the value of the zero of the transfer function of the sigma delta high pass filter, e.g., with the complement of the gain factor.
    • 通过将输入求和节点预设到Σ-Δ调制器,从一开始就从数字滤波器例如Σ-Δ滤波器(特别是从Σ-Δ高通滤波器)中去除DC瞬变。 虽然可以使用任何适当的输入来预设输入求和节点,但是在所公开的实施例中,通过在包含非零预设值的输入和包括输出的正常输入之间切换部分反馈项来预设Σ-Δ高通滤波器 输入求和节点。 基于Σ-Δ高通滤波器的传递函数的零值的值例如以增益因子的补数来选择预设值。