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    • 31. 发明授权
    • Low frequency spectral enhancement system and method
    • 低频谱增强系统和方法
    • US06233549B1
    • 2001-05-15
    • US09199072
    • 1998-11-23
    • Anthony P. MauroGilbert C. Sih
    • Anthony P. MauroGilbert C. Sih
    • G10L1906
    • G10L21/02G10L21/0232
    • A system for enhancing low frequency spectral content of a digitized signal which identifies a fundamental frequency component in the signal and selectively boosts signals within a predetermined range thereof. In the illustrative embodiment, the digitized signal is a frequency domain transformed speech signal. The invention amplifies the low frequency components of the speech signal. The speaker unique fundamental frequency of the speech is computed using pitch delay information and is thus dynamic from frame to frame and also speaker to speaker. This fundamental frequency defines the center point of a gain window which is applied to select frequency components. Only such fundamental frequency components which exhibit a large enough signal to noise ratio have the amplification function applied. Thus, this function can be applied directly following a noise suppression system which has knowledge of the signal quality in each frequency bin. The gain window is ramped up and hanged over to smooth the amplification function between successive frames.
    • 一种用于增强数字化信号的低频谱含量的系统,其识别信号中的基频分量并且选择性地提高其预定范围内的信号。 在说明性实施例中,数字化信号是频域变换的语音信号。 本发明放大语音信号的低频分量。 使用音调延迟信息来计算扬声器独特的基本频率,并且因此是帧间动态的,而扬声器也是扬声器。 该基本频率定义了应用于选择频率分量的增益窗口的中心点。 仅具有足够大的信噪比的基频分量具有应用的放大功能。 因此,该功能可以直接应用于具有每个频率仓中的信号质量知识的噪声抑制系统。 增益窗口上升并挂起,以平滑连续帧之间的放大功能。
    • 32. 发明授权
    • Noise replacement system and method in an echo canceller
    • 回波消除器中的噪声替换系统和方法
    • US5646991A
    • 1997-07-08
    • US607402
    • 1996-02-23
    • Gilbert C. Sih
    • Gilbert C. Sih
    • H04M1/60G10L19/012H04B3/23H04M9/08H04B3/20
    • H04B3/234G10L19/012H04B3/23H04M9/082
    • A noise replacement system and method for providing a synthesized noise replacement signal to an output speech channel in an echo cancellation system. A far-end speech signal transmitted through a communications channel may be passed through an echo channel to produce an echo signal. The echo signal is summed with near-end speech and transmitted back to the far-end in the output speech channel. An echo canceller rejects the echo signal by forming an estimate of the echo signal and subtracting the estimate from the sum of the near-end speech and the echo signal to produce an echo residual signal. When only the far-end speaker is talking, the echo is completely rejected by replacing the echo residual signal with a synthesized noise signal. The noise is synthesized to match the power and spectral characteristics of the actual background noise at the near-end to prevent the far-end speaker from detecting any change in signal characteristics.
    • 一种用于向回声消除系统中的输出语音信道提供合成噪声置换信号的噪声替换系统和方法。 可以通过通信信道发送的远端语音信号通过回波信道以产生回波信号。 回波信号与近端语音求和并传输回输出语音信道中的远端。 回波消除器通过形成回波信号的估计并从近端语音和回波信号的和减去估计值来拒绝回波信号,以产生回波残留信号。 当只有远端扬声器正在说话时,通过用合成噪声信号代替回波残差信号,完全拒绝回波。 合成噪声以匹配近端实际背景噪声的功率和频谱特性,以防止远端扬声器检测信号特性的任何变化。
    • 34. 发明授权
    • Network echo canceller
    • 网络回波消除器
    • US5307405A
    • 1994-04-26
    • US951074
    • 1992-09-25
    • Gilbert C. Sih
    • Gilbert C. Sih
    • H04M1/60G10L19/012H04B3/23H04M9/08H04M1/00
    • H04B3/234G10L19/012H04B3/23H04M9/082
    • An echo canceller and method for cancelling in a return channel signal an echoed receive channel signal where the echoed receive channel signal is combined by an echo channel with an input return channel signal. The echo canceller has a first filter which generates first filter coefficients, generates a first echo estimate signal with the first filter coefficients, and updates the first filter coefficients in response to a first filter control signal. A first summer subtracts the first echo estimate signal from a combined return channel and echo receive channel signal to generate a first echo residual signal. A second filter generates second filter coefficients, generates a second echo estimate signal with the second filter coefficients, and updates the second filter coefficients in response to a second filter control signal. A second summer subtracts the second echo estimate signal from the combined signal to generate a second echo residual signal, and provides upon the return channel the second echo residual signal. A control unit determines from the receive channel signal, the combined signal, and the first and second echo residual signals, one of a plurality of control states wherein a first control state is indicative of a receive channel signal above a first predetermined energy level, wherein when the control unit is in the first control state it generates the first control signal and generates the second control signal when at least one of a first energy ratio of the first echo residual signal and the combined signal and a second energy ratio of the second echo residual signal and the combined signal exceed a predetermined level.
    • 回波消除器和用于在返回信道中消除回波的接收信道信号的方法,其中回波的接收信道信号由回波信道与输入返回信道信号组合。 回波消除器具有产生第一滤波器系数的第一滤波器,产生具有第一滤波器系数的第一回波估计信号,并响应于第一滤波器控制信号来更新第一滤波器系数。 第一个夏季从组合返回信道和回波接收信道信号中减去第一回波估计信号,以产生第一回波残差信号。 第二滤波器产生第二滤波器系数,产生具有第二滤波器系数的第二回波估计信号,并且响应于第二滤波器控制信号来更新第二滤波器系数。 第二个夏天从组合信号中减去第二回波估计信号,以产生第二回波残差信号,并在返回信道上提供第二回波残留信号。 控制单元从接收信道信号,组合信号以及第一和第二回波残余信号中确定多个控制状态中的一个,其中第一控制状态指示高于第一预定能级的接收信道信号,其中 当所述控制单元处于所述第一控制状态时,产生所述第一控制信号,并且当所述第一回波残留信号和所述组合信号的第一能量比和所述第二回波的第二能量比中的至少一个产生所述第二控制信号时, 剩余信号和组合信号超过预定水平。
    • 35. 发明授权
    • Data boundary aware base station assisted position location
    • 数据边界感知基站辅助位置定位
    • US07949351B2
    • 2011-05-24
    • US12277104
    • 2008-11-24
    • Qiuzhen ZouGilbert C. SihInyup Kang
    • Qiuzhen ZouGilbert C. SihInyup Kang
    • H04Q7/20
    • G01S5/0036G01S19/235G01S19/254G01S19/258G01S19/30
    • The present invention is a novel and improved method and apparatus for performing position location in wireless communications system. One embodiment of the invention comprises a method for performing position location on a subscriber unit in a terrestrial wireless telephone system using a set of satellites each transmitting a signal, the terrestrial wireless telephone system having base stations, including the steps of transmitting an aiding message from the base station to the subscriber unit, said aiding message containing information regarding a data boundary for each signal from the set of satellites, applying correlation codes to each signal yielding corresponding correlation data and accumulating said correlation data over an first interval preceding a corresponding data boundary yielding a first accumulation result, and a second interval following said corresponding data boundary yielding a second accumulation result.
    • 本发明是一种用于在无线通信系统中执行位置定位的新颖且改进的方法和装置。 本发明的一个实施例包括一种用于在陆地无线电话系统中的用户单元上执行位置定位的方法,该方法使用一组发射信号的卫星,具有基站的地面无线电话系统包括以下步骤:从 基站到用户单元,所述辅助消息包含关于来自卫星组的每个信号的数据边界的信息,将相关代码应用于产生相应相关数据的每个信号,并在对应的数据边界之前的第一间隔上累加所述相关数据 产生第一累积结果,并且在所述对应数据边界之后的第二间隔产生第二累积结果。
    • 38. 发明授权
    • Variable length instruction decoder
    • 可变长度指令解码器
    • US06425070B1
    • 2002-07-23
    • US09044086
    • 1998-03-18
    • Qiuzhen ZouGilbert C. SihInyup KangQuaeed MotiwalaDeepu JohnLi ZhangHaitao ZhangWay-Shing Lee
    • Qiuzhen ZouGilbert C. SihInyup KangQuaeed MotiwalaDeepu JohnLi ZhangHaitao ZhangWay-Shing Lee
    • G06F9302
    • G06F9/3816G06F9/30014G06F9/30098G06F9/30149G06F9/30152G06F9/3885G06F15/7857
    • The present invention is a novel and improved method and circuit for digital signal processing. One aspect of the invention calls for the use of a variable length instruction set. A portion of the variable length instructions may be stored in adjacent locations within memory space with the beginning and ending of instructions occurring across memory word boundaries. Furthermore, additional aspects of the invention are realized by having instructions contain variable numbers of instruction fragments. Each instruction fragment causes a particular operation, or operations, to be performed allowing multiple operations during each clock cycle. Thus, multiple operations are performed during each clock cycle, reducing the total number of clock cycles necessary to perform a task. The exemplary DSP includes a set of three data buses over which data may be exchanged with a register bank and three data memories. The use of more than two data buses, and especially three data buses, realizes another aspect of the invention, which is significantly reduced bus contention. One embodiment of the invention calls for the data buses to include one wide bus and two narrow buses. The wide bus is coupled to a wide data memory and the two narrow buses are coupled to two narrow data memories. Another aspect of the invention is realized by the use of a register bank that has registers accessible by at least two processing units. This allows multiple operations to be performed on a particular set of data by the multiple processing units, without reading and writing the data to and from a memory. The processing units in the exemplary embodiment of the invention include an arithmetic logic (ALU) and a multiply-accumulate (MAC) unit. When combined with the use of the multiple bus architecture, highly parallel instructions, or both, an additional aspect of the invention is realized where highly pipelined, multi-operation, processing is performed.
    • 本发明是用于数字信号处理的新颖且改进的方法和电路。 本发明的一个方面要求使用可变长度指令集。 可变长度指令的一部分可以存储在存储器空间内的相邻位置,同时跨越存储器字边界的指令的开始和结束。 此外,通过使指令包含可变数量的指令片段来实现本发明的附加方面。 每个指令片段导致执行特定操作或操作,允许在每个时钟周期期间进行多个操作。 因此,在每个时钟周期期间执行多个操作,减少执行任务所需的总时钟周期数。 示例性DSP包括一组三个数据总线,数据可以通过该数据总线与寄存器组和三个数据存储器交换。 使用两条以上的数据总线,特别是三条数据总线,实现了本发明的另一方面,这显着减少了总线竞争。 本发明的一个实施例要求数据总线包括一个宽的总线和两个窄的总线。 宽总线耦合到宽数据存储器,并且两个窄总线耦合到两个窄数据存储器。 通过使用具有可由至少两个处理单元访问的寄存器的寄存器组来实现本发明的另一方面。 这允许通过多个处理单元对特定数据集执行多个操作,而不向存储器读取和写入数据。 本发明的示例性实施例中的处理单元包括算术逻辑(ALU)和乘法累加(MAC)单元。 当结合使用多总线架构,高度并行指令或两者时,实现本发明的另一方面,其中执行高度流水线化,多操作的处理。
    • 39. 发明授权
    • Noise-compensated speech recognition templates
    • 噪声补偿语音识别模板
    • US06381569B1
    • 2002-04-30
    • US09018257
    • 1998-02-04
    • Gilbert C. SihNing Bi
    • Gilbert C. SihNing Bi
    • G10L1520
    • G10L15/20G10L21/0216
    • The speech recognition training unit is modified to store digitized speech samples into a speech database that can be accessed at recognition time. The improved recognition unit comprises a noise analysis, modeling, and synthesis unit which continually analyzes the noise characteristics present in the audio environment and produces an estimated noise signal with similar characteristics. The recognition unit then constructs a noise-compensated template database by adding the estimated noise signal to each of the speech samples in the speech database and performing parameter determination on the resulting sums. This procedure accounts for the presence of noise in the recognition phase by retraining all the templates using an estimated noise signal with similar characteristics as the actual noise signal that corrupted the word to be recognized. This method improves the likelihood of a good template match, which increases the recognition accuracy.
    • 修改语音识别训练单元以将数字化语音样本存储到可在识别时被访问的语音数据库中。 改进的识别单元包括噪声分析,建模和合成单元,其连续分析存在于音频环境中的噪声特性并产生具有相似特性的估计噪声信号。 然后,识别单元通过将估计的噪声信号加到语音数据库中的每个语音样本上并对所得到的和进行参数确定来构建噪声补偿模板数据库。 该过程通过使用具有与损坏要识别的字的实际噪声信号相似的特性的估计噪声信号重新训练所有模板来解决识别阶段中的噪声的存在。 该方法提高了模板匹配的可能性,从而提高了识别精度。