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    • 1. 发明授权
    • Alphanumeric message composing method using telephone keypad
    • 使用电话键盘的字母数字消息编写方法
    • US06052443A
    • 2000-04-18
    • US166739
    • 1998-10-05
    • Surendra PereraLu ChangJian-Cheng HuangRobert J. Schwendeman
    • Surendra PereraLu ChangJian-Cheng HuangRobert J. Schwendeman
    • G06F3/02H04M20060101H04M1/247H04M1/515H04M11/00H04Q7/16
    • G06F3/0237H04M2250/70
    • An interactive method for composing an alphanumeric message by a caller using a telephone keypad includes storing (215) a lexical database (135) from which unigram probabilities, forward conditional probabilities, and backward conditional probabilities for a plurality of words can be recovered; storing a received sequence of key codes (405) representing a sequence in which keys on a telephone style keypad are keyed; generating a word trellis including candidate words (415) derived from the sequence and the lexical database; determining a most likely phrase (420) from the candidate words, the unigram probabilities, forward conditional probabilities, and backward conditional probabilities; generating a most likely message (425) from the most likely phrase and presenting the most likely message to the caller; and confirming that the most likely message is the alphanumeric message (430).
    • 一种用于使用电话键盘由呼叫者构成字母数字消息的交互方法包括存储(215)词法数据库(135),可从中恢复多个单词的单位概率,前向条件概率和反向条件概率; 存储表示键盘键盘上的键的序列的接收的键码序列(405); 生成包括从序列导出的候选词(415)和词汇数据库的单词格; 从所述候选词确定最可能的短语(420),所述单字组概率,前向条件概率和向后条件概率; 从最可能的短语产生最可能的消息(425),并向呼叫者呈现最可能的消息; 并确认最可能的消息是字母数字消息(430)。
    • 4. 发明授权
    • Voice compression by phoneme recognition and communication of phoneme
indexes and voice features
    • 语音压缩由音素识别和通信的音素指标和语音特征
    • US6073094A
    • 2000-06-06
    • US89081
    • 1998-06-02
    • Lu ChangJian-Cheng HuangRobert J. Schwendeman
    • Lu ChangJian-Cheng HuangRobert J. Schwendeman
    • G10L19/00G10L19/14
    • G10L19/0018
    • 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.
    • 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。
    • 5. 发明授权
    • Subband normalization, transformation, and voiceness to recognize
phonemes for text messaging in a radio communication system
    • 子带规范化,转换和语音识别无线电通信系统中文本消息的音素
    • US6163765A
    • 2000-12-19
    • US50184
    • 1998-03-30
    • Oleg AndricLu ChangJian-Cheng HuangArthur Gerald Herkert
    • Oleg AndricLu ChangJian-Cheng HuangArthur Gerald Herkert
    • G10L15/02G10L25/93H04W88/18
    • H04W4/18G10L15/02G10L2015/025G10L2025/937H04W4/14
    • A radio communication system includes a voice recognition system (221) for converting (400) a caller's voice message to a textual speech message. The textual speech message is then transmitted to an intended selective call radio (122). To perform these functions, the radio communication system includes a caller interface circuit (218), a transmitter (116), and a processor (222). To perform voice-to-text conversion, the processor is adapted to cause the caller interface circuit to sample a voice signal generated by the caller during a plurality of frame intervals, and to apply a Fourier transform thereto, thereby generating spectral data. The spectral data is subdivided into a plurality of bands. The spectral envelope of the spectral data is then filtered out to generate filtered spectral data. A Fourier transform is applied thereto to generate an autocorrelation function for each band. From the autocorrelation function of each band, a magnitude is determined, which is representative of the degree of voiceness of each band. The degree of voiceness for each band is then applied to a corresponding plurality of phoneme models, which are used to derive a textual equivalent of speech from the voice signal. The textual equivalent of speech is then transmitted to the selective call radio by way of the transmitter.
    • 无线电通信系统包括用于将呼叫者的语音消息转换(400)到文本语音消息的语音识别系统(221)。 然后将文本语音消息发送到预期的选择呼叫无线电(122)。 为了执行这些功能,无线电通信系统包括呼叫者接口电路(218),发射器(116)和处理器(222)。 为了执行语音到文本转换,处理器适于使呼叫者接口电路在多个帧间隔期间采样由呼叫者产生的语音信号,并对其进行傅里叶变换,由此产生频谱数据。 光谱数据被细分成多个频带。 然后滤出光谱数据的光谱包络以产生滤波的光谱数据。 对其进行傅立叶变换以产生每个频带的自相关函数。 根据每个频带的自相关函数,确定一个幅度,其代表每个频带的声音程度。 然后将每个频带的声音程度应用于相应的多个音素模型,这些音素模型用于从语音信号中导出语音的文本等价物。 然后通过发射机将语音的文本等价物发送到选呼通话。
    • 7. 发明申请
    • AVATAR FOR A PORTABLE DEVICE
    • 用于便携式设备的AVATAR
    • US20090251484A1
    • 2009-10-08
    • US12062098
    • 2008-04-03
    • Ming-Xi ZhaoJian-Cheng Huang
    • Ming-Xi ZhaoJian-Cheng Huang
    • G09G5/02
    • H04M1/72544H04M2250/52
    • 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.
    • 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。
    • 8. 发明授权
    • Digital signal processor for processing voice messages
    • 用于处理语音信息的数字信号处理器
    • US06691081B1
    • 2004-02-10
    • US09560110
    • 2000-04-28
    • Jian-Cheng HuangKenneth D. FinlonFloyd D. Simpson
    • Jian-Cheng HuangKenneth D. FinlonFloyd D. Simpson
    • G10L1106
    • G10L13/0335G10L13/02H04M3/533
    • 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.
    • 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。
    • 9. 发明授权
    • Pitch determiner for a speech analyzer
    • 语音分析仪的音调确定器
    • US6018706A
    • 2000-01-25
    • US999171
    • 1997-12-29
    • Jian-Cheng HuangFloyd SimpsonXiaojun Li
    • Jian-Cheng HuangFloyd SimpsonXiaojun Li
    • G10L11/04G10L19/12G10L7/06
    • G10L25/90G10L19/12G10L19/09G10L21/013
    • 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.
    • 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。
    • 10. 发明授权
    • Method and apparatus for minimal redundancy error detection and
correction of voice spectrum parameters
    • 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
    • US5636231A
    • 1997-06-03
    • US523578
    • 1995-09-05
    • Jian-Cheng HuangXiaojun LiFloyd Simpson
    • Jian-Cheng HuangXiaojun LiFloyd Simpson
    • G10L19/00G10L19/06H03M13/00
    • G10L19/07G10L19/005
    • 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.
    • 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。