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    Method of and apparatus for generating auxiliary information for expediting sparse codebook search
    11.
    发明授权
    Method of and apparatus for generating auxiliary information for expediting sparse codebook search 失效
    用于产生辅助信息的方法和装置用于进行稀疏的代码簿搜索

    公开(公告)号：US5195137A

    公开(公告)日：1993-03-16

    申请号：US646122

    申请日：1991-01-28

    Applicant: Kumar Swaminathan

    Inventor： Kumar Swaminathan

    IPC: G06T9/00 ,  H03M7/30

    CPC classification number: G06T9/008 ,  H03M7/3082

    Abstract: In many applications involving the coding and processing of speech signals the relevant applicable codebook is one which may be termed a sparse codebook. That is, the majority of elements in the codebook are zero valued. The searching of such a sparse codebook is accelerated in accord with the present invention by generating auxiliary information defining the sparse nature of the codebok and using this information to assist and speed up searches of the codebook.In a particular method of searching the calculation of the distance between a target vector and a stored codebook vector is enhanced by use of a distortion metric derived from energy terms and correlation terms of the codebook entries. Calculation of these energy and correlation terms is speeded up by exploiting the sparseness of the codebook entries. The non-zero elements (NZE) of the space codebook are each identified and are defined by their offset from a reference point.

    In-band transmission of TTY/TTD signals for systems employing low bit-rate voice compression
    12.
    发明授权
    In-band transmission of TTY/TTD signals for systems employing low bit-rate voice compression 有权
    对采用低比特率语音压缩的系统进行TTY / TTD信号的带内传输

    公开(公告)号：US06961320B1

    公开(公告)日：2005-11-01

    申请号：US09669283

    申请日：2000-09-26

    Applicant: Kumar Swaminathan ,  Udaya Bhaskar

    Inventor： Kumar Swaminathan ,  Udaya Bhaskar

    IPC: H04L5/22 ,  H04M3/42 ,  H04M11/00 ,  H04M11/06

    CPC classification number: H04M3/42391 ,  H04M11/066 ,  H04M2207/18

    Abstract: A method, system, and software product for transmitting TTY/TDD signals in a system employing low bit-rate voice compression are disclosed. The method includes receiving an input signal and generating a teletypewriter (TTY) indicator signal from the input signal. Whether or not the input signal is a TTY signal including a TTY character, is determined based on the TTY indicator signal. A TTY packet including the TTY character of the TTY signal is constructed and transmitted if the input signal is determined to be a TTY signal. A method, system, and software product for receiving and decoding TTY/TDD signal is also disclosed.

    Abstract translation: 公开了一种在采用低比特率语音压缩的系统中传输TTY / TDD信号的方法，系统和软件产品。 该方法包括接收输入信号并从输入信号产生电传打字机（TTY）指示符信号。 是否基于TTY指示符信号确定输入信号是否包括TTY字符的TTY信号。 如果输入信号被确定为TTY信号，则构造并发送包括TTY信号的TTY字符的TTY分组。 还公开了用于接收和解码TTY / TDD信号的方法，系统和软件产品。

    Parallel/pipeline VLSI architecture for a low-delay CELP coder/decoder
    14.
    发明授权
    Parallel/pipeline VLSI architecture for a low-delay CELP coder/decoder 有权
    用于低延迟CELP编码器/解码器的并行/流水线VLSI架构

    公开(公告)号：US06314393B1

    公开(公告)日：2001-11-06

    申请号：US09270918

    申请日：1999-03-16

    Applicant: Yue-Peng Zheng ,  Shvetal K. Patel ,  Kumar Swaminathan

    Inventor： Yue-Peng Zheng ,  Shvetal K. Patel ,  Kumar Swaminathan

    IPC: G10L1912

    CPC classification number: G10L19/16 ,  G10L19/12

    Abstract: An integrated circuit for processing a speech signal in accordance with a CELP standard includes a plurality of processing elements coupled to a data bus in parallel. Each processing element includes a multiplier and an accumulator. The integrated circuit further includes an auxiliary processing element, which is also coupled to the data bus and has a division unit and a comparator. The plurality of processing elements and the auxiliary processing element are also coupled in a pipeline formation.

    Abstract translation: 用于根据CELP标准处理语音信号的集成电路包括并行耦合到数据总线的多个处理单元。 每个处理元件包括一个乘法器和一个累加器。 集成电路还包括辅助处理元件，其还耦合到数据总线，并具有除法单元和比较器。 多个处理元件和辅助处理元件也以管道结构耦合。

    Speech mode based multi-stage vector quantizer
    15.
    发明授权
    Speech mode based multi-stage vector quantizer 失效
    基于语音模式的多级矢量量化器

    公开(公告)号：US5966688A

    公开(公告)日：1999-10-12

    申请号：US958143

    申请日：1997-10-28

    Applicant: Srinivas Nandkumar ,  Kumar Swaminathan

    Inventor： Srinivas Nandkumar ,  Kumar Swaminathan

    IPC: G10L11/06 ,  G10L19/00 ,  G10L19/06 ,  G10L3/02

    CPC classification number: G10L19/07 ,  G10L25/93

    Abstract: A speech mode based multi-stage vector quantizer is disclosed which quantizes and encodes line spectral frequency (LSF) vectors that were obtained by transforming the short-term predictor filter coefficients in a speech codec that utilizes linear predictive techniques. The quantizer includes a mode classifier that classifies each speech frame of a speech signal as being associated with one of a voiced, spectrally stationary (Mode A) speech frame, a voiced, spectrally non-stationary (Mode B) speech frame and an unvoiced (Mode C) speech frame. A converter converts each speech frame of the speech signal into an LSF vector and an LSF vector quantizer includes a 12-bit, two-stage, backward predictive vector encoder that encodes the Mode A speech frames and a 22 bit, four-stage backward predictive vector encoder that encodes the Mode 13 and the Mode C speech frames.

    Abstract translation: 公开了一种基于语音模式的多级矢量量化器，其对通过使用线性预测技术的语音编解码器中的短期预测器滤波器系数进行变换而获得的线谱频率（LSF）矢量进行量化和编码。 量化器包括模式分类器，其将语音信号的每个语音帧分类为与有声，频谱平稳（模式A）语音帧，有声，频谱非平稳（模式B）语音帧和无声（ 模式C）语音帧。 A转换器将语音信号的每个语音帧转换为LSF向量，并且LSF向量量化器包括对模A语音帧进行编码的12位两级反向预测向量编码器和22位四级后向预测 编码模式13和模式C语音帧的矢量编码器。

    Mode-specific method and apparatus for encoding signals containing speech
    17.
    发明授权
    Mode-specific method and apparatus for encoding signals containing speech 失效
    用于编码包含语音的信号的模式特定方法和装置

    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。

    High quality low bit rate celp-based speech codec
    18.
    发明授权
    High quality low bit rate celp-based speech codec 失效
    高质量低比特率基于celp的语音编解码器

    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。

    Improving sub-band coding of speech at low bit rates by adding residual speech energy signals to sub-bands
    19.
    发明授权
    Improving sub-band coding of speech at low bit rates by adding residual speech energy signals to sub-bands 失效
    通过向子带添加残留语音能量信号，以低比特率改进语音的子带编码

    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。

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