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    • 4. 发明申请
    • SYSTEM AND METHOD FOR LONG RANGE AND SHORT RANGE DATA COMPRESSION
    • 用于长范围和短距离数据压缩的系统和方法
    • US20130018932A1
    • 2013-01-17
    • US13180969
    • 2011-07-12
    • Udaya BhaskarChi-Jiun Su
    • Udaya BhaskarChi-Jiun Su
    • G06F17/10
    • H04L65/60H03M7/3088H03M7/4006H04L69/04
    • A system and method are provided for use with streaming blocks of data, each of the streaming blocks of data including a number bits of data. The system includes a first compressor and a second compressor. The first compressor can receive and store a number n blocks of the streaming blocks of data, can receive and store a block of data to be compressed of the streaming blocks of data, can compress consecutive bits within the block of data to be compressed based on the n blocks of the streaming blocks of data, can output a match descriptor and a literal segment. The match descriptor is based on the compressed consecutive bits. The literal segment is based on a remainder of the number of bits of the data to be compressed not including the consecutive bits. The second compressor can compress the literal segment and can output a compressed data block including the match descriptor and a compressed string of data based on the compressed literal segment.
    • 提供了一种用于流数据块的系统和方法,每个流数据块包括数位数据。 该系统包括第一压缩机和第二压缩机。 第一压缩器可以接收和存储流数据块的数个n个块,可以接收和存储要压缩数据流数据块的数据块,可以基于压缩的数据块中的连续位来压缩 数据流块的n个块可以输出匹配描述符和文本段。 匹配描述符基于压缩的连续位。 文字片段基于要压缩的数据的比特数的剩余部分不包括连续的比特。 第二压缩器可以压缩字面部分并且可以输出包括匹配描述符的压缩数据块和基于压缩文字段的压缩数据串。
    • 5. 发明授权
    • Voicing measure for a speech CODEC system
    • 语音CODEC系统的语音测量
    • US07013269B1
    • 2006-03-14
    • US10073406
    • 2002-02-13
    • Udaya BhaskarKumar Swaminathan
    • Udaya BhaskarKumar Swaminathan
    • G10L19/02
    • G10L19/097G10L25/93
    • 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 providing 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 also provides a pitch contour within the predetermined intervals. A voice activity detector adapted to process the LP parameters and the open loop pitch contour over the predetermined intervals is also provided as well as a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following functions: 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 invervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and provide a voicing measure where the voicing measure characterizes a degree of vocing 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. The voicing measure is provided for the purpose of regenerating a PW phase at a decoder; and providing improved quantization of the PW magnitude at an encoder. The voicing measure is encoded jointly with a PW nonstationarity measure vector using a spectrally weighted vector quantizer having a codebook partioned based on a voiced and unvoiced mode.
    • 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编解码器系统,其包括线性预测(LP)前端,其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,并使用 以计算LP残差信号。 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也在预定间隔内提供音调轮廓。 还提供了适于在预定间隔上处理LP参数和开环音调轮廓的语音活动检测器以及响应于LP残差信号和音调轮廓的信号处理器,并且适于执行以下功能:提取原型 来自LP残差的波形(PW)和开环节距轮廓线,用于在预定的反相中的多个相等子间隔; 通过PW的增益值对PW进行归一化; 编码PW的大小; 并且提供发声测量,其中所述发声测量表征所述输入语音信号的声音程度,并且从与所述预定间隔上的所述信号的周期度相关的若干输入参数导出。 提供发声措施是为了在解码器处再生PW相; 并且在编码器处提供对PW幅度的改进的量化。 发声测量与PW非平稳测量向量一起编码,其使用具有基于有声和无声模式分组的码本的频谱加权矢量量化器。
    • 7. 发明授权
    • Method and system for reducing echo and noise in a vehicle passenger compartment environment
    • 减轻车辆乘客室环境中回波和噪音的方法和系统
    • US08625775B2
    • 2014-01-07
    • US12852341
    • 2010-08-06
    • Udaya BhaskarPeng Lee
    • Udaya BhaskarPeng Lee
    • H04M9/08G10K11/16A61F11/06
    • H04M9/082
    • An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.
    • 通信装置中的回波消除算法基于相对于接收信号的功率电平的背景噪声信号功率电平和相对于接收信号的输出的回波估计的功率电平来初始化在自适应回波滤波器中使用的步长值 回波消除器。 然后,该算法调整步长值。 一个方面基于在一个或多个干扰信号频率处检测大的快速傅立叶变换值来调整步长。 另一方面估计残余回波能量以调整估计的回波能量,其然后用于设置双方通话标志,如果发送信号具有比估计的回波信号多得多的功率。 另一方面将发送信号功率与发送信号功率的抽取版本进行比较,如果前者超过预定量,则设置双方通话标志。
    • 8. 发明授权
    • Prototype waveform magnitude quantization for a frequency domain interpolative speech codec system
    • 用于频域内插语音编解码系统的原型波形幅度量化
    • US06996523B1
    • 2006-02-07
    • US10073128
    • 2002-02-13
    • Udaya BhaskarKumar Swaminathan
    • Udaya BhaskarKumar Swaminathan
    • G10L19/10
    • G10L19/097G10L19/032
    • 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).
    • 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。
    • 10. 发明申请
    • METHOD AND SYSTEM FOR REDUCING ECHO AND NOISE IN A VEHICLE PASSENGER COMPARTMENT ENVIRONMENT
    • 车辆乘客舱室环境中降低噪声和噪音的方法与系统
    • US20110033059A1
    • 2011-02-10
    • US12852341
    • 2010-08-06
    • Udaya BhaskarPeng Lee
    • Udaya BhaskarPeng Lee
    • G10K11/16
    • H04M9/082
    • An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.
    • 通信装置中的回波消除算法基于相对于接收信号的功率电平的背景噪声信号功率电平和相对于接收信号的输出的回波估计的功率电平来初始化在自适应回波滤波器中使用的步长值 回波消除器。 然后,该算法调整步长值。 一个方面基于在一个或多个干扰信号频率处检测大的快速傅立叶变换值来调整步长。 另一方面估计残余回波能量以调整估计的回波能量,其然后用于设置双方通话标志,如果发送信号具有比估计的回波信号多得多的功率。 另一方面将发送信号功率与发送信号功率的抽取版本进行比较,如果前者超过预定量,则设置双方通话标志。