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    • 2. 发明授权
    • User-Customized, low bit-rate speech vocoding method and communication
unit for use therewith
    • 用户定制的低比特率语音声码方法和与其一起使用的通信单元
    • US5774856A
    • 1998-06-30
    • US537583
    • 1995-10-02
    • William Joe HaberGeorge Thomas KronckeWilliam George SchmidtMartin John O'Sullivan
    • William Joe HaberGeorge Thomas KronckeWilliam George SchmidtMartin John O'Sullivan
    • G10L17/00G10L19/14G10L3/02
    • G10L25/00G10L19/00G10L17/00
    • A voice encoding method �and apparatus initialize! initializes (160) transmit and receive vocoders operated within communication units (12, 42) with a user-unique speech characteristic model (SCM) table (320) and an input stimulus table (340). The SCM table (320) and input stimulus table (340) are created during a training task (120) and stored (144) in a user information card (360), within a communication unit (60), or at a control facility (90). During call setup, the tables are exchanged (180) between the transmit and receive vocoders and the user's speech is encoded (200) using the SCM table (320) and input stimulus table (340). During encoding (200), either compressed table entries that most closely match the input speech, or indexes that identify the closest table entries are sent (220) in a bitstream to the receive vocoder. The SCM table (320) and input stimulus table (340) can be updated (240) during or after the call.
    • 语音编码方法[和装置初始化]利用用户唯一的语音特征模型(SCM)表(320)和输入激励表(340)来初始化(160)发送和接收在通信单元(12,42)内操作的声码器。 SCM表(320)和输入激励表(340)在训练任务(120)期间创建并存储(144)在用户信息卡(360)中,在通信单元(60)内,或在控制设备 90)。 在呼叫建立期间,在发送和接收声码器之间交换表(180),并且使用SCM表(320)和输入激励表(340)对用户的语音进行编码(200)。 在编码(200)期间,将最接近匹配输入语音的压缩表条目或识别最接近的表条目的索引在比特流中发送(220)到接收声码器。 在呼叫期间或之后,SCM表(320)和输入激励表(340)可被更新(240)。
    • 3. 发明授权
    • Digital speech interpolation method and apparatus
    • 数字语音插值方法和装置
    • US5754536A
    • 1998-05-19
    • US550319
    • 1995-10-30
    • William George Schmidt
    • William George Schmidt
    • H04J3/14H04J3/17H04J4/00H04J3/16
    • H04J3/14H04J3/172H04J4/00
    • The method and apparatus of the present invention uses DSI with a TDMA/FDMA communication protocol. For each speech frame detected by a speech detector (110), a communication unit (CU) (108) sends (440) a speech detected indicator (SDI) (208) to a switching facility (SF) (138). In response, the SF (138) allocates an uplink reuse unit (260) which identifies a carrier frequency (301-360) and timeslot (190-193) which the CU (108) should use to transmit a traffic burst containing a compressed speech frame. The SF (138) allocates (506) the uplink reuse unit (260) from a pool of available uplink reuse units and, if necessary, allocates a downlink reuse unit (262) from a pool of available downlink reuse units. A message (220) describing the reuse units (260, 262) is transmitted (508) to the CU (108) which then transmits (450) and receives (480) traffic bursts during the allocated reuse units (260, 262).
    • 本发明的方法和装置使用具有TDMA / FDMA通信协议的DSI。 对于由语音检测器(110)检测到的每个语音帧,通信单元(CU)(108)向语音检测器(108)发送(440)语音检测指示符(SDI)(208)(138)。 作为响应,SF(138)分配上行链路重用单元(260),其标识CU(108)应用于发送包含压缩语音的业务突发的载波频率(301-360)和时隙(190-193) 帧。 SF(138)从可用上行链路重用单元池中分配(506)上行链路重用单元(260),如果需要,从可用下行链路重用单元池中分配下行链路重用单元(262)。 向CU(108)发送描述重用单元(260,262)的消息(220)(508),然后在分配的重用单元(260,262)期间发送(450)并接收(480)业务突发。
    • 5. 发明授权
    • Transmitter system and method of operation therefor
    • 变送器系统及其操作方法
    • US6157681A
    • 2000-12-05
    • US55395
    • 1998-04-06
    • Brian Michael DanielKenneth Maynard PetersonWilliam George Schmidt
    • Brian Michael DanielKenneth Maynard PetersonWilliam George Schmidt
    • H01Q3/26H03D5/00H04L27/18H04L27/34H04L27/36H03C7/02H04B1/02H04L27/04
    • H04L27/18H01Q3/26H03D5/00H04L27/34
    • A transmitter system (18) which transmits a communication signal (14) through a phased-array antenna (10) uses a signal processor (26) and a multiplicity of direct modulators (30). Each direct modulator (30) drives a single element (12) of the antenna (10). Each direct modulator (30) receives a digital baseband phase point data signal (62) from a digital data stream (28). The digital data stream (28) also conveys digital beam formation data (60). Each direct modulator (30) digitally combines the baseband phase point data (62) with the beam formation data (60) to produce digital streams that modulate an RF carrier signal (34). The digital data stream (28) is generated by a phased-array signal processor (26) which processes user data bits (86) received from any number of input signals (22) and processes beam signals (24). In one embodiment, the signal processor (26) calculates phase point data (62) representative of all input signals (22).
    • 通过相控阵天线(10)发送通信信号(14)的发射机系统(18)使用信号处理器(26)和多个直接调制器(30)。 每个直接调制器(30)驱动天线(10)的单个元件(12)。 每个直接调制器(30)从数字数据流(28)接收数字基带相位点数据信号(62)。 数字数据流(28)还传送数字波束形成数据(60)。 每个直接调制器(30)将基带相位点数据(62)与波束形成数据(60)进行数字组合,以产生调制RF载波信号(34)的数字流。 数字数据流(28)由处理从任意数量的输入信号(22)接收的用户数据位(86)并处理波束信号(24)的相控阵列信号处理器(26)产生。 在一个实施例中,信号处理器(26)计算表示所有输入信号(22)的相位点数据(62)。
    • 6. 发明授权
    • Method and apparatus for transmitting user-customized high-quality,
low-bit-rate speech
    • 发送用户定制的高质量,低比特率语音的方法和装置
    • US6094628A
    • 2000-07-25
    • US28111
    • 1998-02-23
    • William Joe HaberGeorge Thomas KronckeWilliam George Schmidt
    • William Joe HaberGeorge Thomas KronckeWilliam George Schmidt
    • G10L19/00G10L25/00G10L3/02
    • G10L19/00G10L25/00
    • A method and apparatus for improving the quality and transmission rates of speech is presented. Upon connection of a call with a receiving terminal, a communication unit (12, 26, 28, 42, 57, 54, 60) reads a dynamic user-specific speech characteristics model (SCM) table and user-specific input stimulus table and sends them to an appropriate point in the connection path with the receiving terminal. As normal voice conversation begins, the user's speech is collected into speech frames. The speech frames are compared to input stimuli entries in the user-specific input stimulus table, and are used to calculate SCMs which are compared to dynamic user-specific SCM table entries in the dynamic user-specific SCM table to generate an encoded bit stream. Simultaneously, speech characteristics statistics are collected and analyzed in view of multiple available generic SCMs to update and improve the dynamic user-specific SCM table during the progress of the call to closely track changes in the user's voice.
    • 提出了一种用于提高语音质量和传输速率的方法和装置。 在通话与接收终端连接时,通信单元(12,26,28,42,57,54,60)读取动态用户特定语音特征模型(SCM)表和用户专用输入激励表,并发送 它们到与接收终端的连接路径中的适当点。 当正常语音对话开始时,用户的语音被收集到语音帧中。 将语音帧与用户专用输入激励表中的输入刺激条目进行比较,并且用于计算与动态用户专用SCM表中的动态用户特定SCM表条目进行比较以生成编码比特流的SCM。 同时,根据多个可用的通用SCM收集和分析语音特征统计信息,以在通话过程中更新和改进动态用户特定的SCM表,以密切跟踪用户语音的变化。