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    • 1. 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。
    • 2. 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。
    • 3. 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。
    • 4. 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。
    • 5. 发明授权
    • Method and apparatus for transferring low bit rate digital voice messages using incremental messages
    • 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
    • US06772126B1
    • 2004-08-03
    • US09410006
    • 1999-09-30
    • Floyd SimpsonJian-Cheng HuangSunil SatyamurtiKenneth FinlonRobert Schwendeman
    • Floyd SimpsonJian-Cheng HuangSunil SatyamurtiKenneth FinlonRobert Schwendeman
    • G10L2104
    • G10L19/24
    • 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.
    • 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。
    • 7. 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。
    • 8. 发明授权
    • Very low bit rate voice messaging system using variable rate backward
search interpolation processing
    • 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
    • US5682462A
    • 1997-10-28
    • US528033
    • 1995-09-14
    • Jian-Cheng HuangFloyd SimpsonXiaojun Li
    • Jian-Cheng HuangFloyd SimpsonXiaojun Li
    • G10L19/00G10L19/06G10L9/00
    • G10L19/06
    • 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.
    • 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。