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公开(公告)号：US06691081B1
公开(公告)日：2004-02-10
申请号：US09560110
申请日：2000-04-28
IPC分类号： G10L1106
CPC分类号： G10L13/0335 , G10L13/02 , H04M3/533
摘要： A digital signal processor for processing data including voice messaging data that may have both voiced and unvoiced speech components utilizes computer routines stored in a memory used by the digital signal processor. The computer routines programmed provide for control of at least a portion of a selective call receiver; receiving and decoding data received at the selective call receiver; comparing the addresses received at the selective call receiver with addresses stored in a memory location coupled to the digital signal processor; controlling voicing including both voiced and unvoiced speech components; and generating a pitch wave using an inverse discrete Fourier Transform and resample the pitch wave to provide a time domain voiced speech component.
摘要翻译： 用于处理包括语音消息传送数据的数字信号处理器可以具有有声和无声话音分量两者利用存储在由数字信号处理器使用的存储器中的计算机程序。 编程的计算机程序提供用于控制选呼接收机的至少一部分; 接收和解码在选呼接收机处接收的数据; 将在选呼接收机处接收的地址与存储在耦合到数字信号处理器的存储器位置中的地址进行比较; 控制声音，包括有声和无声语音成分; 以及使用逆离散傅里叶变换产生音调波并重新采样音调波以提供时域有声语音分量。
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2.发明授权
公开(公告)号：US6064955A
公开(公告)日：2000-05-16
申请号：US58924
申请日：1998-04-13
IPC分类号： G10L13/02 , G10L13/033 , H04M3/533 , G10L11/06
CPC分类号： G10L13/0335 , G10L13/02 , H04M3/533
摘要： A MBE synthesizer (2200) for generating speech from information received by a receiver (114) includes a voiced signal generator (2280) for generating voiced signal components in the time domain using an IDFT in a pitch wave generator (2210) and and a pitch wave resampler (2232) and an unvoiced signal generator (2290) for generating unvoiced signal components in the time domain. The MBE synthesizer also includes a voicing processor (2218) responsive to band voicing flags within the excitation information for controlling selection of a voiced spectral component or an unvoiced spectral component from a harmonic amplitude spectrum.
摘要翻译： 一种用于从由接收器（114）接收的信息产生语音的MBE合成器（2200）包括用于使用音调波发生器（2210）中的IDFT在时域中产生有声信号分量的语音信号发生器（2280） 波重采样器（2232）和用于在时域中产生清音信号分量的清音信号发生器（2290）。 MBE合成器还包括响应于激励信息内的频带发声标志的声控处理器（2218），用于控制来自谐波振幅频谱的有声频谱分量或无声频谱分量的选择。
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3.发明授权MBE synthesizer utilizing a nonlinear voicing processor for very low bit rate voice messaging 失效MBE合成器利用非线性语音处理器实现非常低的比特率语音消息
公开(公告)号：US5806038A
公开(公告)日：1998-09-08
申请号：US600582
申请日：1996-02-13
申请人： Jian-Cheng Huang , Floyd D. Simpson , Xiaojun Li
发明人： Jian-Cheng Huang , Floyd D. Simpson , Xiaojun Li
CPC分类号： G10L19/0212 , G10L19/10
摘要： A MBE (Multi-Band Excitation) synthesizer (116) generates excitation components from information received by a receiver (2004). The information received includes spectral information representing a segment of speech. The MBE synthesizer (116) includes an excitation generator (2241) and a nonlinear voicing processor (2211). The excitation generator (2241) generates voiced excitation components and unvoiced excitation components. The nonlinear voicing processor (2211) is responsive to the spectral information and controls a selection of the excitation components from the voiced excitation components and the unvoiced excitation components.
摘要翻译： MBE（多频带激励）合成器（116）从由接收机接收的信息（2004）产生激励分量。 所接收的信息包括表示语音段的频谱信息。 MBE合成器（116）包括激励发生器（2241）和非线性发声处理器（2211）。 励磁发生器（2241）产生有声激励分量和无声激励分量。 非线性发声处理器（2211）响应于频谱信息并且控制来自有声激励分量和无声激励分量的激励分量的选择。
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公开(公告)号：US20090251484A1
公开(公告)日：2009-10-08
申请号：US12062098
申请日：2008-04-03
申请人： Ming-Xi Zhao , Jian-Cheng Huang
发明人： Ming-Xi Zhao , Jian-Cheng Huang
IPC分类号： G09G5/02
CPC分类号： H04M1/72544 , H04M2250/52
摘要： A portable device comprises a data storage for storing avatar data defining a user avatar. The user avatar is formed by a plurality of visual objects. The portable device further comprises a camera for capturing an image. A visual characteristic processor is arranged to determine a first visual characteristic from the image and an avatar processor is arranged to set an object visual characteristic of an object of the plurality of visual objects in response to the first visual characteristic. The invention may allow improved customization of user avatars. For example, a color of an element of a user avatar may be adapted to a color of a real-life object simply by a user taking a picture thereof.
摘要翻译： 便携式设备包括用于存储定义用户头像的头像数据的数据存储器。 用户头像由多个视觉对象形成。 便携式设备还包括用于捕获图像的相机。 视觉特征处理器被布置成从图像中确定第一视觉特征，并且化身处理器被布置成响应于第一视觉特性来设置多个视觉对象的对象的对象视觉特征。 本发明可以允许改进用户化身的定制。 例如，用户头像的元素的颜色可以简单地由用户拍摄其图片而适应于现实生活中的对象的颜色。
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公开(公告)号：US6018706A
公开(公告)日：2000-01-25
申请号：US999171
申请日：1997-12-29
申请人： Jian-Cheng Huang , Floyd Simpson , Xiaojun Li
发明人： Jian-Cheng Huang , Floyd Simpson , Xiaojun Li
CPC分类号： G10L25/90 , G10L19/12 , G10L19/09 , G10L21/013
摘要： A pitch determiner (414) for use with a speech analyzer includes a pitch function generator (414) which generates a plurality of pitch components representing a pitch function for one or more sequential segments of speech. which are represented by a predetermined number of digitized speech samples. A pitch enhancer (1116) enhances the pitch function of a current segment of speech utilizing the pitch function of one or more sequential segments of speech to generate a plurality of enhanced pitch components. A pitch detector (1118) detects the pitch of the current segment of speech by determining the pitch of an enhanced pitch component having a largest amplitude of the plurality of enhanced pitch components.
摘要翻译： 与语音分析器一起使用的音调确定器（414）包括音调函数发生器（414），其生成表示用于一个或多个连续语音段的音调函数的多个音调分量。 其由预定数量的数字化语音样本表示。 音调增强器（1116）利用一个或多个连续语音段的音调函数来增强当前语音段的音调函数，以产生多个增强音调分量。 音高检测器（1118）通过确定具有多个增强音调分量的最大振幅的增强音调分量的音调来检测当前语音段的音高。
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6.发明授权Method and apparatus for minimal redundancy error detection and correction of voice spectrum parameters 失效用于语音频谱参数的最小冗余错误检测和校正的方法和装置
公开(公告)号：US5636231A
公开(公告)日：1997-06-03
申请号：US523578
申请日：1995-09-05
申请人： Jian-Cheng Huang , Xiaojun Li , Floyd Simpson
发明人： Jian-Cheng Huang , Xiaojun Li , Floyd Simpson
CPC分类号： G10L19/07 , G10L19/005
摘要： Error detection and correction of a received message, such as a digitized voice message is achieved by generating (318) interpolated vectors for each error vector corresponding to a codebook index in a sequence of codebook indexes representing parameters of portions of the message. A plurality of error corrected candidate vectors for the vector corresponding to the codebook index in error, are generated (322,324,326) by flipping one bit in a sequence of bits representing the codebook index in error. The error corrected candidate vector which has a minimal difference from its corresponding interpolated vector is used (338) to replace the error vector. In the case of digital voice, the vectors are spectral vectors which represent spectral information for a time sample of a voice message. An ordering property of vector components is exploited to detect errors in a received codebook index without parity bits.
摘要翻译： 通过对表示信息部分参数的代码簿索引序列中的码本索引生成每个误差向量的内插向量来实现对诸如数字化语音消息的接收消息的错误检测和校正。 通过在代表码本索引的位的序列中翻转一位来产生用于与错误码本索引相对应的矢量的多个纠错候选向量（322,324,326）。 使用与其对应的内插向量具有最小差异的误差校正候选向量（338）来替换误差向量。 在数字语音的情况下，矢量是表示语音消息的时间采样的频谱信息的频谱矢量。 利用矢量分量的排序属性来检测接收到的码本索引中没有奇偶校验位的错误。
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7.发明授权Method and apparatus for transferring low bit rate digital voice messages using incremental messages 失效用于使用增量消息传送低比特率数字语音消息的方法和装置公开(公告)号：US06772126B1
公开(公告)日：2004-08-03
申请号：US09410006
申请日：1999-09-30
IPC分类号： G10L2104
CPC分类号： G10L19/24
摘要： A system controller (106) is for transferring a low bit rate digital voice message. The system controller generates from an analog voice signal representing the voice message a set of speech model parameters, and generates a first derived set of speech model parameters from a first subset of the set of speech model parameters, the first derived set encoding the voice signal at a second voice quality and second vocoder rate that are less, respectively, than a first voice quality and vocoder rate. The system controller transmits (3610) the low bit rate-digital voice message comprising the first derived set of speech model parameters to a communication receiver (114). The communication receiver requests (3640) an incremental message when the quality of the voice message is unsatisfactory. The system controller generates and transmits (3555, 3650) an incremental message-and the communication receiver uses (3660) the incremental message to generate a higher quality voice message.
摘要翻译： 系统控制器（106）用于传送低比特率数字话音消息。 系统控制器从表示语音消息的模拟语音信号产生一组语音模型参数，并且从语音模型参数集合的第一子集生成第一导出的语音模型参数集，编码语音信号的第一导出集合 分别具有比第一语音质量和声码率更小的第二语音质量和第二声码器速率。 系统控制器将包括第一导出的语音模型参数集合的低比特率数字话音消息（3610）发送到通信接收机（114）。 当语音消息的质量不令人满意时，通信接收器请求（3640）增量消息。 系统控制器生成并发送增量消息（3555,3650），通信接收器使用增量消息（3660）生成更高质量的语音消息。
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8.发明授权Method and apparatus for dynamic segmentation of a low bit rate digital voice message 失效低比特率数字语音消息的动态分割方法和装置公开(公告)号：US06418405B1
公开(公告)日：2002-07-09
申请号：US09410140
申请日：1999-09-30
IPC分类号： G10L1900
CPC分类号： G10L15/04 , G10L19/0018
摘要： A system controller (106) includes a speech encoder (107) that dynamically segments frames of a low bit rate digital voice message. Speech model parameters have been generated in a sequence of frames. The speech model parameters include quantized speech spectral parameter vectors. The speech encoder selects (1820) a first quantized speech spectral parameter vector as a current anchor vector, selects (1820, 1830) a second quantized speech spectral parameter vector located a predetermined number of frames (LMAX) from the current anchor vector as a target speech parameter vector, and perturbs (1840) the target speech parameter vector to derive a plurality (K) of perturbed speech parameter vectors.
摘要翻译： 系统控制器（106）包括动态地分段低比特率数字语音消息的帧的语音编码器（107）。 语音模型参数已经在一系列帧中生成。 语音模型参数包括量化语音频谱参数向量。 语音编码器选择（1820）作为当前锚矢量的第一量化语音频谱参数矢量，选择（1820,1830）从位于当前锚矢量的预定数量帧（LMAX）的第二量化语音频谱参数矢量作为目标 语音参数向量，并扰动（1840）目标语音参数向量，以导出多个（K）扰动语音参数向量。
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9.发明授权Voice compression by phoneme recognition and communication of phoneme indexes and voice features 失效语音压缩由音素识别和通信的音素指标和语音特征
公开(公告)号：US6073094A
公开(公告)日：2000-06-06
申请号：US89081
申请日：1998-06-02
CPC分类号： G10L19/0018
摘要： A communication system includes a transmitter for transmitting messages to a plurality of receiving devices of the communication system, and a processing system. The processing system is adapted to convert a caller's voice message to a sequence of phonemes whereby the caller's voice message is intended for a receiving device. To accomplish the conversion, steps of Fourier transform, spectral subdivision, envelope filtering autocorrelation function determination of each subdivision, and voiceness determination for each subdivision are performed. The processing system is further adapted to generate a sequence of phoneme indexes and voice features corresponding to the sequence of phonemes, and to cause the transmitter to transmit the sequence of phoneme indexes to the receiving device for generating a voice signal representative of the caller's voice message. The voice features can include spectral features, average energy, duration, and pitch to improve the quality of the voice signal. The receiving device can be a selective call radio.
摘要翻译： 通信系统包括用于向通信系统的多个接收设备发送消息的发射机和处理系统。 处理系统适于将呼叫者的语音消息转换成一系列音素，由此呼叫者的语音消息用于接收设备。 为了完成转换，执行傅立叶变换，频谱细分，每个细分的包络滤波自相关函数确定步骤和每个细分的声音确定步骤。 该处理系统进一步适用于产生与音素序列相对应的音素索引和语音特征序列，并使发射机将音素索引序列发射到接收装置，以产生代表呼叫者语音消息的语音信号 。 语音特征可以包括频谱特征，平均能量，持续时间和音高以提高语音信号的质量。 接收设备可以是选呼通话。
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10.发明授权Very low bit rate voice messaging system using variable rate backward search interpolation processing 失效使用可变速率反向搜索插值处理的非常低比特率语音消息系统
公开(公告)号：US5682462A
公开(公告)日：1997-10-28
申请号：US528033
申请日：1995-09-14
申请人： Jian-Cheng Huang , Floyd Simpson , Xiaojun Li
发明人： Jian-Cheng Huang , Floyd Simpson , Xiaojun Li
CPC分类号： G10L19/06
摘要： A method and apparatus is provided for a low bit rate speech transmission. Speech spectral parameter vectors are generated from a voice message and stored in a sequence of speech spectral parameter vectors within a speech spectral parameter matrix. A first index identifying a first speech parameter template corresponding to a first speech spectral parameter vector of the sequence of speech spectral parameter vectors is transmitted. A subsequent speech spectral parameter vector of the sequence is selected and a subsequent speech parameter template is determined having a subsequent index. One or more intervening interpolated speech parameter templates are interpolated between the first speech parameter template and the subsequent speech parameter template. The one or more intervening speech spectral parameter vectors are compared to the corresponding one or more intervening interpolated speech parameter templates to derive a distance. The subsequent index is transmitted when the distance derived is less than or equal to a predetermined distance.
摘要翻译： 提供了一种用于低比特率语音传输的方法和装置。 语音频谱参数矢量从语音消息生成并存储在语音频谱参数矩阵内的语音频谱参数矢量序列中。 发送识别对应于语音频谱参数矢量序列的第一语音频谱参数向量的第一语音参数模板的第一索引。 选择该序列的后续语音频谱参数矢量，并且确定随后的语音参数模板具有后续索引。 在第一语音参数模板和随后的语音参数模板之间插入一个或多个插入的内插语音参数模板。 将一个或多个中间语音频谱参数矢量与相应的一个或多个插入的内插语音参数模板进行比较以导出距离。 当所导出的距离小于或等于预定距离时，传送随后的索引。
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