METHOD AND APPARATUS FOR RECEIVING AN ENCODED SPEECH SIGNAL
    13.
    发明申请
    METHOD AND APPARATUS FOR RECEIVING AN ENCODED SPEECH SIGNAL 失效
    用于接收编码的语音信号的方法和装置

    公开(公告)号:US20110270608A1

    公开(公告)日:2011-11-03

    申请号:US12853861

    申请日:2010-08-10

    Applicant: Daniel Lin

    Inventor: Daniel Lin

    CPC classification number: G10L19/12 G10L2019/0005 G10L2019/0007

    Abstract: A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A decoder receives encoded speech including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index. An innovation codevector and an adaptive codevector are determined and scaled. An excitation sequence is generated. Reconstructed speech is then output based on the excitation sequence and LSP index.

    Abstract translation: 在无线通信系统中处理语音的方法和装置使用CELP语音编码信号。 解码器接收包括码索引,码索引增益,音调滞后,音调增益和线谱对(LSP)索引的编码语音。 确定和缩放创新代码向量和自适应代码向量。 产生激励序列。 然后根据激励序列和LSP索引输出重构语音。

    Method and device for encoding speech using open-loop pitch analysis
    14.
    发明授权
    Method and device for encoding speech using open-loop pitch analysis 失效
    使用开环音调分析对语音进行编码的方法和装置

    公开(公告)号:US07599832B2

    公开(公告)日:2009-10-06

    申请号:US11363807

    申请日:2006-02-28

    CPC classification number: G10L19/20 G10L19/06 G10L19/09 G10L19/10 G10L25/90

    Abstract: The present invention is a synthetic speech encoding device that produces a synthetic speech signal which closely matches an actual speech signal. The actual speech signal is digitized, and excitation pulses are selected by minimizing the error between the actual and synthetic speech signals. The preferred pattern of excitation pulses needed to produce the synthetic speech signal is obtained by using an excitation pattern containing a multiplicity of weighted pulses at timed positions. The selection of the location and amplitude of each excitation pulse is obtained by minimizing an error criterion between the synthetic speech signal and the actual speech signal. The error criterion function incorporates a perceptual weighting filter which shapes the error spectrum.

    Abstract translation: 本发明是一种合成语音编码装置,其产生与实际语音信号紧密匹配的合成语音信号。 实际语音信号被数字化,并且通过最小化实际和合成语音信号之间的误差来选择激励脉冲。 通过使用在定时位置处包含多个加权脉冲的激励模式来获得产生合成语音信号所需的激励脉冲的优选模式。 通过最小化合成语音信号和实际语音信号之间的误差标准来获得每个激励脉冲的位置和振幅的选择。 错误标准函数包含形成误差谱的感知加权滤波器。

    Peer-to-peer mobile instant messaging method and device
    16.
    发明申请
    Peer-to-peer mobile instant messaging method and device 有权
    对等移动即时通讯方式和设备

    公开(公告)号:US20050220134A1

    公开(公告)日:2005-10-06

    申请号:US10817994

    申请日:2004-04-05

    Applicant: Daniel Lin

    Inventor: Daniel Lin

    CPC classification number: H04L51/04 H04L51/38

    Abstract: A technique is provided for establishing peer-to-peer session-based instant messaging between mobile devices without the need for using an instant messaging registration or log-in server to provide presence information. Session-based instant messaging communications between mobile devices are established by transmitting necessary address information through page-based messaging services that utilize the underlying digital mobile network databases and services to resolve the identification and location of the mobile devices.

    Abstract translation: 提供了一种用于在移动设备之间建立基于对等会话的即时消息传递的技术,而不需要使用即时消息登记或登录服务器来提供呈现信息。 通过使用基础数字移动网络数据库和服务的基于页面的消息传送服务传送必要的地址信息来建立移动设备之间的基于会话的即时消息通信,以解决移动设备的识别和位置。

    Peer-to-peer mobile data transfer method and device
    17.
    发明申请
    Peer-to-peer mobile data transfer method and device 有权
    对等移动数据传输方式和设备

    公开(公告)号:US20050220041A1

    公开(公告)日:2005-10-06

    申请号:US11042620

    申请日:2005-01-24

    Applicant: Daniel Lin

    Inventor: Daniel Lin

    CPC classification number: H04L51/38 H04W8/26 H04W68/00 H04W76/14 H04W92/18

    Abstract: A technique is provided for a direct data transfer session, including for multimedia content, between mobile devices without the need for using a separate multimedia server to store multimedia content. Direct data transfer sessions between mobile devices are established by transmitting necessary address information through page-based messaging services that utilize the underlying digital mobile network databases and services to resolve the identification and location of the mobile devices.

    Abstract translation: 提供了用于在移动设备之间的用于直接数据传输会话的技术,包括用于多媒体内容,而不需要使用单独的多媒体服务器来存储多媒体内容。 通过使用底层数字移动网络数据库和服务的基于页面的消息传送服务传送必要的地址信息来建立移动设备之间的直接数据传输会话,以解决移动设备的识别和位置。

    Multiple impulse excitation speech encoder and decoder

    公开(公告)号:US06385577B1

    公开(公告)日:2002-05-07

    申请号:US09805634

    申请日:2001-03-14

    Abstract: A version of a speech signal and an output of a pitch synthesis filter and a linear predictive all-pole (LPC) filter is received. A system impulse response is produced based on in part the received pitch synthesis filter and LPC output. An excitation pulse location is determined so that the determined location minimizes an error between the speech signal version and the system impulse response. The speech signal is encoded with a representation of the determined location.

    Multiple impulse excitation speech encoder and decoder
    19.
    发明授权
    Multiple impulse excitation speech encoder and decoder 失效
    多脉冲激励语音编码器和解码器

    公开(公告)号:US06006174A

    公开(公告)日:1999-12-21

    申请号:US950658

    申请日:1997-10-15

    CPC classification number: G10L19/20 G10L19/06 G10L19/10 G10L25/90 G10L19/09

    Abstract: The generation of multipulse excitation codes by digitizing an original speech, partitioning the digitized signal into a number of samples, pre-emphasizing the samples, producing linear predictive reflection coefficients from said samples, quantizing these reflection coefficients, converting the quantized reflection coefficients to spectral coefficients and subjecting the spectral coefficients to pitch analysis to obtain a spectral residual signal.

    Abstract translation: 通过数字化原始语音来产生多脉冲激励码,将数字化信号划分成多个样本,预先强调样本,从所述样本产生线性预测反射系数,量化这些反射系数,将量化的反射系数转换成频谱系数 并对频谱系数进行音调分析以获得频谱残差信号。

    Robust vector quantization of line spectral frequencies
    20.
    发明授权
    Robust vector quantization of line spectral frequencies 失效
    线谱频率的鲁棒矢量量化

    公开(公告)号:US5651026A

    公开(公告)日:1997-07-22

    申请号:US495148

    申请日:1995-06-27

    CPC classification number: G10L19/07 G06Q10/0875 G10L25/24

    Abstract: A line spectral frequency (LSF) vector quantizer, having particular application in digital cellular networks (DCN), is provided for code excited linear predictive (CELP) speech encoders. The LSF vector quantizer is efficient in terms of bits employed, robust and effective in terms of performance across speakers and handsets, moderate in terms of complexity, and accommodates effective and simple built-in transmission error detection schemes. The LSF vector quantizer employs a minimum number of bits, is of moderate complexity and incorporates built-in error detection capability in order to combat transmission errors. The LSF vector quantizer classifies unquantized line spectral frequencies into four categories, employing different vector quantization tables for each category. Each quantization table is optimized for particular types of vectors. For each category, three split vector codebooks are used with a simplified error measure to find three candidate split quantized vectors. The three sets of three split vectors are combined to produce as many as 27 vectors from each category. The quantizer then makes a final selection of optimal category using a more complex error measure to achieve the robust performance across speakers and handsets. Split vector quantization follows a two stage constrained search procedure that results in an ordered set of quantized line spectral frequencies that is "close" to the unquantized set with moderate complexity within each category. Effective and simple transmission error detection schemes at the receiver are made possible by the split nature of the vector quantization and the constrained search procedure. Only twenty-six bits are required to encode ten line spectral frequencies.

    Abstract translation: 为码激励线性预测(CELP)语音编码器提供了一种在数字蜂窝网络(DCN)中具有特殊应用的线谱频率(LSF)矢量量化器。 LSF矢量量化器在所采用的比特方面是有效的,在扬声器和手机上的性能方面是稳健和有效的,在复杂性方面适中,并且适应有效和简单的内置传输错误检测方案。 LSF矢量量化器采用最小数量的位,具有中等复杂度,并且具有内置的错误检测能力,以抵御传输错误。 LSF矢量量化器将未量化的线谱频率分为四类,每类使用不同的矢量量化表。 每个量化表针对特定类型的向量进行了优化。 对于每个类别,使用三个分离矢量码本与简化误差测量来找到三个候选分割量化矢量。 将三组三个分割矢量组合起来,从每个类别产生多达27个矢量。 然后量化器使用更复杂的误差测量来最终选择最佳类别,以实现扬声器和手机的强大性能。 分割矢量量化遵循两阶段约束搜索过程,其导致在每个类别内具有中等复杂度的未量化集合的有序集合的量化线谱频率。 通过矢量量化和约束搜索过程的分割性质,可以实现接收机的有效和简单的传输错误检测方案。 只需要二十六位来编码十个线谱频率。

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