公开(公告)号：US5758256A

公开(公告)日：1998-05-26

申请号：US483426

申请日：1995-06-07

Applicant: Kirk H. Berry ,  Debabrata Chowdhury ,  Shrirang Jangi ,  Kumar Swaminathan

Inventor： Kirk H. Berry ,  Debabrata Chowdhury ,  Shrirang Jangi ,  Kumar Swaminathan

IPC: G10L19/00 ,  H04W88/18 ,  H04M11/00 ,  H04Q7/00

CPC classification number: H04W88/181 ,  G10L19/012

Abstract: A method of transporting speech information over a wireless cellular communications system is provided. By determining the existence and compatibility of the destination port with the origination port in a given telephone call, the present invention is capable of using only one compression step and one decompression step. Accordingly, voice signal degradation and delay associated with multiple compression/decompression steps may be reduced.

Abstract translation: 提供了一种通过无线蜂窝通信系统传送语音信息的方法。 通过在给定的电话呼叫中确定目的地端口与起始端口的存在和兼容性，本发明能够仅使用一个压缩步骤和一个解压缩步骤。 因此，可以减少与多个压缩/解压缩步骤相关联的语音信号衰减和延迟。

公开(公告)号：US5596676A

公开(公告)日：1997-01-21

申请号：US540637

申请日：1995-10-11

Applicant: Kumar Swaminathan ,  Kalyan Ganesan ,  Prabhat K. Gupta

Inventor： Kumar Swaminathan ,  Kalyan Ganesan ,  Prabhat K. Gupta

IPC: G10L11/04 ,  G10L11/06 ,  G10L19/00 ,  G10L19/12 ,  G10L19/14 ,  G10L9/12 ,  G10L9/14

CPC classification number: G10L19/012 ,  G10L19/12 ,  G10L19/26 ,  G10L25/90 ,  G10L25/09 ,  G10L25/18 ,  G10L25/24 ,  G10L25/93

Abstract: A method for encoding a signal that includes a speech component is described. First and second linear prediction windows of a frame are analyzed to generate sets of filter coefficients. First and second pitch analysis windows of the frame are analyzed to generate pitch estimates. The frame is classified in one of at least two modes, e.g. voiced, unvoiced and noise modes, based, for example, on pitch stationarity, short-term level gradient or zero crossing rate. Then the frame is encoded using the filter coefficients and pitch estimates in a particular manner depending upon the mode determination for the frame, preferably employing CELP based encoding algorithms.

Abstract translation: 描述了一种用于编码包括语音分量的信号的方法。 分析帧的第一和第二线性预测窗口以生成滤波器系数集合。 分析帧的第一和第二音调分析窗口以产生音调估计。 该帧被分类为至少两种模式之一，例如， 例如，基于音调稳定性，短期电平梯度或零交叉率的有声，无声和噪声模式。 然后，根据帧的模式确定，优选使用基于CELP的编码算法，以特定方式使用滤波器系数和音调估计来对帧进行编码。

公开(公告)号：US5495555A

公开(公告)日：1996-02-27

申请号：US905992

申请日：1992-06-25

Applicant: Kumar Swaminathan

Inventor： Kumar Swaminathan

IPC: G10L19/08 ,  G10L11/04 ,  G10L11/06 ,  G10L19/00 ,  G10L19/04 ,  G10L19/12 ,  G10L19/14 ,  G10L3/02

CPC classification number: G10L19/26 ,  G10L19/12 ,  G10L25/90 ,  G10L25/93

Abstract: Code excited linear prediction (CELP) is performed using two voiced and unvoiced sets of windows, each set is used both for linear prediction and pitch determination. The accompanying degradation in voice quality is comparable to the IS54 standard 8.0 Kbps voice coder employed in U.S. digital cellular systems. This is accomplished by using the same parametric model used in traditional CELP coders but determining, quantizing, encoding, and updating these parameters differently. The low bit rate speech decoder is like most CELP decoders except that it operates in two modes depending on the received mode bit. Both pitch prefiltering and global postfiltering are employed for enhancement of the synthesized speech. In addition, built-in error detection and error recovery schemes are used that help mitigate the effects of any uncorrectable transmission errors.

Abstract translation: 代码激励线性预测（CELP）是使用两个有声和无声的窗口组来执行的，每组都用于线性预测和音调确定。 伴随的语音质量下降与美国数字蜂窝系统中使用的IS54标准8.0Kbps语音编码器相当。 这通过使用与传统CELP编码器中使用的相同的参数模型来实现，但是以不同的方式确定，量化，编码和更新这些参数。 低比特率语音解码器像大多数CELP解码器一样，除了它根据接收模式位在两种模式下操作。 采用两种音调预滤波和全局后置滤波来增强合成语音。 此外，使用内置的错误检测和错误恢复方案，有助于减轻任何不可纠正的传输错误的影响。

公开(公告)号：US4956871A

公开(公告)日：1990-09-11

申请号：US252250

申请日：1988-09-30

Applicant: Kumar Swaminathan

Inventor： Kumar Swaminathan

IPC: H03M1/12 ,  G10L19/02 ,  G10L19/12 ,  H03M7/30 ,  H04B1/66 ,  H04B14/04

CPC classification number: H04B1/667

Abstract: A sub-band speech coding arrangement divides the speech spectrum into sub-bands and allocates bits to encode the time frame interval samples of each sub-band responsive to the speech energies of the sub-bands. The sub-band samples are quantized according to the sub-band energy bit allocation and the time frame quantized samples and speech energy signals are coded. A signal representative of the residual difference between the each time frame interval speech sample of the sub-band and the corresponding quantized speech sample of the sub-band is generated. The quality of the sub-band coded signal is improved by selecting the sub-bands with the largest residual differences, producing a vector signal from the sequence of residual difference signals of each selected sub-band, and matching the sub-band vector signal to one of a set of stored Gaussian codebook entries to generate a reduced bit code for the selected vector signal. The coded time frame interval quantized signals, speech energy signals and reduced bit codes for the selected residual differences are combined to form a multiplexed stream for the speech pattern of the time frame interval.

Abstract translation: 子带语音编码装置将语音频谱划分为子频带，并且分配比特以响应于子频带的语音能量对每个子频带的时间间隔采样进行编码。 根据子带能量比特分配对子带样本进行量化，并对时间帧量化样本和语音能量信号进行编码。 产生表示子带的每个时间间隔语音样本与子带的对应的量化语音样本之间的残差的信号。 通过选择具有最大残差的子带来改善子带编码信号的质量，从每个选择的子带的残差差信号的序列产生矢量信号，并将子带向量信号与 一组存储的高斯码本条目中的一个，以生成所选择的矢量信号的缩减比特码。 对所选择的残差进行编码的时间间隔量化信号，语音能量信号和降低的比特码被组合以形成用于时间间隔的语音模式的多路复用流。

公开(公告)号：US09249225B2

公开(公告)日：2016-02-02

申请号：US13699944

申请日：2010-12-10

Applicant: John R. Ohlfest ,  Karen Himmel Ohlfest ,  Jayanth Panyam ,  Suresh Kumar Swaminathan ,  Daniel A. Vallera

Inventor： John R. Ohlfest ,  Jayanth Panyam ,  Suresh Kumar Swaminathan ,  Daniel A. Vallera

IPC: C07K16/28 ,  A61K39/00

CPC classification number: C07K16/22 ,  A61K2039/505 ,  A61K2039/507 ,  C07K16/2863 ,  C07K16/2896 ,  C07K16/30 ,  C07K16/3015 ,  C07K16/303 ,  C07K16/3046 ,  C07K2317/622 ,  C07K2317/73 ,  C07K2319/21 ,  C07K2319/42 ,  C07K2319/55 ,  G01N2333/70596

Abstract: Disclosed herein are a monoclonal antibody that specifically binds to human CD 133 and single-chain variable fragments thereof. Also disclosed herein is a hybridoma that produces the monoclonal antibody that specifically binds to human CD133.

Abstract translation: 本文公开了特异性结合人CD133及其单链可变片段的单克隆抗体。 本文还公开了产生特异性结合人CD133的单克隆抗体的杂交瘤。

公开(公告)号：US06996523B1

公开(公告)日：2006-02-07

申请号：US10073128

申请日：2002-02-13

Applicant: Udaya Bhaskar ,  Kumar Swaminathan

Inventor： Udaya Bhaskar ,  Kumar Swaminathan

IPC: G10L19/10

CPC classification number: G10L19/097 ,  G10L19/032

Abstract: A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and directly quantize the PW in a magnitude domain without further decomposition of the PW into complex components, where the direct quantization is performed by a hierarchical quantization method based on a voicing classification using fixed dimension vector quantizers (VQ's).

Abstract translation: 提供了一种系统和方法，其采用用于语音的低比特率编码的频域内插CODEC系统，其包括适于处理提供经过预定间隔量化和编码的LP参数的输入信号的线性预测（LP）前端，以及 用于计算LP残差信号。 还提供了适于处理LP残差信号的开环音调估计器，音调量化器和音调内插器，并且在预定间隔内提供音调轮廓。 还提供了响应于LP残留信号和音调轮廓的信号处理器，并且适于执行以下操作：提供语音测量，其中语音测量表征输入语音信号的发音程度，并且从几个输入参数导出， 与预定间隔的信号的周期度相关; 从所述LP残差和所述开环节距轮廓中提取所述预定间隔内的多个相等子间隔的原型波形（PW）; 通过PW的增益值对PW进行归一化; 编码PW的大小; 并且在幅度域中直接量化PW，而不会将PW进一步分解成复分量，其中通过使用固定维度矢量量化器（VQ's）的基于语音分类的分层量化方法来执行直接量化。

公开(公告)号：US06453289B1

公开(公告)日：2002-09-17

申请号：US09361015

申请日：1999-07-23

Applicant: Filiz Basbug Ertem ,  Srinivas Nandkumar ,  Kumar Swaminathan

Inventor： Filiz Basbug Ertem ,  Srinivas Nandkumar ,  Kumar Swaminathan

IPC: G10L2102

CPC classification number: G10L25/78 ,  G10L21/0208

Abstract: An improved noise reduction algorithm is provided, as well as a voice activity detector, for use in a voice communication system. The voice activity detector allows for a reliable estimate of noise and enhancement of noise reduction. The noise reduction algorithm and voice activity detector can be implemented integrally in an encoder or applied independently to speech coding application. The voice activity detector employs line spectral frequencies and enhanced input speech which has undergone noise reduction to generate a voice activity flag. The noise reduction algorithm employs a smooth gain function determined from a smoothed noise spectral estimate and smoothed input noisy speech spectra. The gain function is smoothed both across frequency and time in an adaptive manner based on the estimate of the signal-to-noise ratio. The gain function is used for spectral amplitude enhancement to obtain a reduced noise speech signal. Smoothing employs critical frequency bands corresponding to the human auditory system. Swirl reduction is performed to improve overall human perception of decoded speech.

Abstract translation: 提供了一种改进的降噪算法，以及用于语音通信系统中的语音活动检测器。 语音活动检测器允许噪声的可靠估计和噪声降低的增强。 噪声降低算法和语音活动检测器可以在编码器中一体地实现或独立地应用于语音编码应用。 语音活动检测器采用经过降噪的线谱频率和增强输入语音以产生语音活动标志。 噪声降低算法采用从平滑噪声谱估计和平滑输入噪声语音谱确定的平滑增益函数。 基于信噪比的估计，以自适应方式在频率和时间上平滑增益功能。 增益函数用于频谱振幅增强，以获得降噪噪声语音信号。 平滑采用对应于人类听觉系统的临界频带。 进行旋转减少以改善对解码语音的整体人感知。

公开(公告)号：US5666370A

公开(公告)日：1997-09-09

申请号：US591127

申请日：1996-01-25

Applicant: Kalyan Ganesan ,  Kumar Swaminathan ,  Prabhat Gupta ,  P. Vijay Kumar

Inventor： Kalyan Ganesan ,  Kumar Swaminathan ,  Prabhat Gupta ,  P. Vijay Kumar

IPC: H03M13/29 ,  H03M13/35 ,  H04L1/00 ,  H04L1/20 ,  G06F11/08

CPC classification number: H04L1/0065 ,  H03M13/09 ,  H03M13/21 ,  H03M13/27 ,  H03M13/29 ,  H03M13/2903 ,  H03M13/2927 ,  H03M13/35 ,  H04L1/0054 ,  H04L1/007 ,  H04L1/201 ,  H04L1/208

Abstract: An improved error control coding scheme is implemented in low bit rate coders in order to improve their performance in the presence of transmission errors typical of the digital cellular channel. The error control coding scheme exploits the nonlinear block codes (NBCs) for purposes of tailoring those codes to a fading channel in order to provide superior error protection to the compressed half rate speech data. For a half rate speech codec assumed to have a frame size of 40 ms, the speech encoder puts out a fixed number of bits per 40 ms. These bits are divided into three distinct classes, referred to as Class 1, Class 2 and Class 3 bits. A subset of the Class 1 bits are further protected by a CRC for error detection purposes. The Class 1 bits and the CRC bits are encoded by a rate 1/2 Nordstrom Robinson code with codeword length of 16. The Class 2 bits are encoded by a punctured version of the Nordstrom Robinson code. It has an effective rate of 8/14 with a codeword length 14. The Class 3 bits are left unprotected. The coded Class 1 plus CRC bits, coded Class 2 bits, and the Class 3 bits are mixed in an interleaving array of size 16.times.17 and interleaved over two slots in a manner that optimally divides each codeword between the two slots. At the receiver the coded Class 1 plus CRC bits, coded Class 2 bits, and Class 3 bits are extracted after de-interleaving.

Abstract translation: 在低比特率编码器中实现改进的误差控制编码方案，以便在存在数字蜂窝信道典型的传输错误的情况下提高它们的性能。 错误控制编码方案利用非线性块码（NBC），以便将这些码定制到衰落信道，以便为压缩的半速率语音数据提供优良的错误保护。 对于假定帧大小为40ms的半速率语音编解码器，语音编码器每40ms提出固定数量的位。 这些位被分为三个不同的类，称为1类，2类和3类。 Class 1位的一个子集进一步受到CRC的保护，以便进行错误检测。 1比特和CRC比特由码字长度为16的速率+ E，fra 1/2 + EE Nordstrom Robinson码编码。2类比特由Nordstrom Robinson码的穿孔版本编码。 它的有效速率为+ E，带有码字长度为14的8/14 + EE。3类比特未被保护。 经编码的1类加CRC比特，2类编码和3类比特混合在大小为16×17的交织阵列中，并且以两个时隙之间的每个码字最佳分割的方式在两个时隙上进行交织。 在接收器处，解码后提取编码的1类加CRC比特，2类编码和3类比特。