Factorial packing method and apparatus for information coding
    1.
    发明专利
    Factorial packing method and apparatus for information coding 审中-公开
    工厂包装方法和信息编码设备

    公开(公告)号:JP2011050070A

    公开(公告)日:2011-03-10

    申请号:JP2010223479

    申请日:2010-10-01

    CPC classification number: G10L19/10

    Abstract: PROBLEM TO BE SOLVED: To provide an improved method and apparatus for allowing efficient coding of a plurality of pulse position tracks.
    SOLUTION: A speech coder takes advantage of the fact that any given pulse combination can be uniquely described by the following four properties: number of degenerate pulses, signs of pulses, positions of pulses, and pulse magnitudes. In accordance with the invention, a four-stage iterative classification of the pulse combinations, where each stage groups the pulse combinations by one of these four properties, is performed. The process starts with the number of pulses and then determines the total number of possible sign combinations, pulse position combinations, and pulse magnitude combinations.
    COPYRIGHT: (C)2011,JPO&INPIT

    Abstract translation: 要解决的问题:提供一种用于允许多个脉冲位置轨道的有效编码的改进的方法和装置。 解决方案:语音编码器利用以下事实:任何给定脉冲组合可以通过以下四个特性唯一描述:简并脉冲数,脉冲符号,脉冲位置和脉冲幅度。 根据本发明,执行脉冲组合的四阶段迭代分类,其中每个阶段通过这四个属性之一来组合脉冲组合。 该过程以脉冲数开始,然后确定可能的符号组合,脉冲位置组合和脉冲幅度组合的总数。 版权所有(C)2011,JPO&INPIT

    A SYSTEM AND METHOD FOR PROVIDING SPLIT VECTOR QUANTIZATION DATA CODING
    2.
    发明申请
    A SYSTEM AND METHOD FOR PROVIDING SPLIT VECTOR QUANTIZATION DATA CODING 审中-公开
    一种用于提供分散矢量量化数据编码的系统和方法

    公开(公告)号:WO9941736A3

    公开(公告)日:1999-10-21

    申请号:PCT/US9902431

    申请日:1999-02-04

    Applicant: MOTOROLA INC

    CPC classification number: G10L19/04 G10L19/06 G10L19/12 H03M7/30

    Abstract: A method and system for providing split vector quantization for use in determining constrained ordered set values, such as line spectrum pair parameters to determine spectral parameters in a data compression system, utilizes multiple codebooks (22a-22c) containing delta coded constrained ordered set values that are normalized to an upper and lower bound. An LSP reconstructor (34) reconstructs received spectral parameters to decode data, such as speech, based on the normalized delta quantization data of line spectrum pair parameters obtained from the split vector reconstruction codebooks (22a-22c). The LSP reconstructor (34) dynamically generates line spectrum pair parameters based on the normalized delta quantization data. In another embodiment, instead of storing the absolute value of the line spectrum pair parameters in segmented codebooks, the combination of at least two absolute value vectors and at least one normalized delta quantization vector is used for spectral quantization.

    Abstract translation: 一种用于提供用于确定约束有序集合值(例如用于确定数据压缩系统中的频谱参数的线谱对参数)的分割矢量量化的方法和系统利用包含增量编码约束有序集合值的多个码本(22a-22c) 被归一化为上限和下限。 基于从分割矢量重建码本(22a-22c)获得的线谱对参数的归一化的量化数据,LSP重构器(34)重建接收的频谱参数以对诸如语音的数据进行解码。 LSP重建器(34)基于归一化的量化数据动态地生成线路频谱对参数。 在另一个实施例中,代替在分段码本中存储线谱对参数的绝对值,而不是将至少两个绝对值向量和至少一个归一化的量化矢量的组合用于频谱量化。

    METHOD AND APPARATUS FOR SUPPRESSING ACOUSTIC BACKGROUND NOISE IN A COMMUNICATION SYSTEM
    3.
    发明公开
    METHOD AND APPARATUS FOR SUPPRESSING ACOUSTIC BACKGROUND NOISE IN A COMMUNICATION SYSTEM 审中-公开
    方法和设备在通信系统抑制剂背景噪声

    公开(公告)号:EP1256112A4

    公开(公告)日:2005-09-07

    申请号:EP00975568

    申请日:2000-11-02

    Applicant: MOTOROLA INC

    CPC classification number: G10L21/0208

    Abstract: A noise suppression system implemented in communication system provides an improved level of quality during severe signal-to-noise ratio (SNR) conditions. The noise suppression system, inter alia, incorporates a frequency domain comb-filtering (289) technique which supplements a traditional spectral noise suppression method. The invention includes a real cepstrum generator (285) for an input signal (285) G(k) to produce a likely voiced speech pitch lag component and converting a result to frequency domain to obtain a comb-filter function (290) C(k), and applying input signal (291) G(k) to comb-filter function (290) C(k) to produce a signal (293) G"(k) to be used for noise suppression. This prevents high frequency components from being unnecessarily attenuated, thereby reducing muffling effects of prior art comb-filters.

    METHOD AND SYSTEM FOR CONTROLLING AN ENCODING RATE IN A VARIABLE RATE COMMUNICATION SYSTEM
    4.
    发明公开
    METHOD AND SYSTEM FOR CONTROLLING AN ENCODING RATE IN A VARIABLE RATE COMMUNICATION SYSTEM 失效
    方法和装置用于控制可变速率通信系统的编码速率

    公开(公告)号:EP1008244A4

    公开(公告)日:2002-10-28

    申请号:EP98920367

    申请日:1998-05-08

    Applicant: MOTOROLA INC

    CPC classification number: H04W28/22 H04B7/2659 H04J3/1688 H04W88/181

    Abstract: A method and system for controlling an encoding rate in a communication system utilizes feedforward rate information (48) and/or rate desirability information (50) sent from each of a plurality of variable rate vocoders (34) to a communication link output controller (24), such as a network arbitor. The communication link output controller (24) then sends a feedback control signal (32) to a selected variable rate vocoder to change the encoding rate of the selected variable rate vocoder to facilitate re-encoding of the speech packet when a bottleneck is detected. In another embodiment, the network arbitor (134) may additionally and independently modify speech packet data when it determines that a bottleneck may occur. The network arbitor (134) also communicates a packet modification control signal (132) for the variable rate vocoder that generated the dropped packet so that the corresponding variable rate vocoder can adjust its filter states to maintain convergence.

    7.
    发明专利
    未知

    公开(公告)号:DE19580846C1

    公开(公告)日:2000-11-16

    申请号:DE19580846

    申请日:1995-05-26

    Applicant: MOTOROLA INC

    Abstract: A method and apparatus are provided for echo canceling which includes double-talk immunity. The echo canceller (15) according to the present invention includes a double-talk detector (17) which uses an adaptive double-talk detection threshold (214) and an accurate estimation of reference energy (106) to detect double-talk. The echo canceller further provides for correcting partial echo canceller adaptive filter vector divergence occurring prior to double-talk detection by maintaining and switching to an auxiliary echo canceller filter with known "good" filter coefficients (58, 60, 62).

    8.
    发明专利
    未知

    公开(公告)号:BR9811630A

    公开(公告)日:2000-10-03

    申请号:BR9811630

    申请日:1998-05-08

    Applicant: MOTOROLA INC

    Abstract: A method and system for controlling speech encoding in a communication system utilizes feedback information such as packet modification control data (154) sent from a communication link output controller, such as a network arbitor (142). The network arbitor (142) sends the packet modification control data (154) to a selected vocoder (146) to change the filter states of the selected vocoder (146) when the network arbitor (142) modifies an output speech packet communicated of a communication link (20), to facilitate improved convergence of a speech encoder and a speech decoder, such as a mobile subscriber unit.

    Method and apparatuus for suppressing noise in a communication system

    公开(公告)号:GB2313266B

    公开(公告)日:2000-01-26

    申请号:GB9713727

    申请日:1996-09-04

    Applicant: MOTOROLA INC

    Abstract: A noise suppression system implemented in communication system provides an improved update decision during instances of sudden increase in background noise level. The noise suppression system, inter alia, generates an update by continually monitoring the deviation of spectral energy and forcing an update based on a predetermined threshold criterion. The spectral energy deviation is determined by utilizing an element which has the past values of the power spectral components exponentially weighted. The exponential weighting is a function of the current input energy, which means the higher the input signal energy the longer the exponential window. Conversely, the lower the signal energy the shorter the exponential window. The noise suppression system also inhibits a forced update during periods of continuous, non-stationary input signals (such as "music-on-hold").

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