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    • 1. 发明授权
    • Full-duplex hands-free transparency circuit and method therefor
    • 全双工免提透明电路及其方法
    • US06799062B1
    • 2004-09-28
    • US09692333
    • 2000-10-19
    • James B. PiketChristopher W. SpringfieldWilliam C. Yip
    • James B. PiketChristopher W. SpringfieldWilliam C. Yip
    • H04B320
    • H04M9/082
    • A bi-directional hands-free communication device includes a microphone for transmitting a signal along a transmit path and a speaker receiving a signal transmitted along a receive path and outputting a corresponding output signal. An echo canceller, positioned in the transmit path and the receive path, cancels echo signals induced by the microphone from the speaker and outputs a corresponding cancelled signal along the transmit path, and a transparency circuit distributes state-dependent additional loss derived from the noise floor margin to the transmit path and the receive path to reduce residual echo signals output from the echo canceller. The transparency circuit measures a noise floor and inserts an artificial noise signal to the transmit path, and optionally to the receive path, at a predetermined level in relation to the measured noise floor, and dynamically adjusts the speaker to compensate for changing environmental conditions by dividing a range of an expected ambient noise power into adjacent consecutive bins, and controlling a volume of the speaker responsive to ambient noise changes only when measured noise power moves into an adjacent bin.
    • 双向免提通信装置包括用于沿发送路径发送信号的麦克风和接收沿着接收路径发送的信号的扬声器,并输出相应的输出信号。 定位在发射路径和接收路径中的回波消除器消除了麦克风从扬声器引起的回波信号,并沿着发射路径输出相应的取消信号,透明电路分配从噪声基底导出的依赖于状态的附加损耗 发送路径和接收路径的余量以减少从回波消除器输出的残余回波信号。 透明电路测量本底噪声,并将噪声信号插入发射路径,并可选地接收到接收路径,以相对于测量的本底噪声为预定水平,并动态调整扬声器,以通过划分来补偿变化的环境条件 预期的环境噪声功率的范围到相邻的连续箱中,并且仅当测量的噪声功率移动到相邻的箱中时,才响应于环境噪声改变扬声器的音量。
    • 2. 发明授权
    • Automatic gain control for an adaptive finite impulse response and method therefore
    • 自适应有限脉冲响应的自动增益控制和方法
    • US07142665B2
    • 2006-11-28
    • US10893034
    • 2004-07-16
    • David L. BarronWilliam C. YipSean S. You
    • David L. BarronWilliam C. YipSean S. You
    • H04B3/23
    • H04M9/082
    • Methods and apparatus are provided for an echo cancellation system. The echo cancellation system comprises an automatic gain control (AGC), a first scalar, a second scalar, a signal summing stage, and an adaptive filter. The AGC is responsive to a first signal in a reference path and a second signal in a near end path. The signal summing stage is between the first and second scalar in the near end path and the first and second scalars are responsive to the AGC. The adaptive filter is responsive to the first signal and provides a third signal to the signal summing stage that corresponds to an echo signal and is subtracted from a signal in the near end path. The second scalar has a scale rate that is the inverse of the scale rate of the first scalar such that unity scaling occurs on the near end path.
    • 为回波消除系统提供了方法和装置。 回波消除系统包括自动增益控制(AGC),第一标量,第二标量,信号求和级和自适应滤波器。 AGC响应于参考路径中的第一信号和近端路径中的第二信号。 信号求和级位于近端路径中的第一和第二标量之间,第一和第二标量响应于AGC。 自适应滤波器响应于第一信号,并向对应于回波信号的信号求和级提供第三信号,并从近端路径中的信号中减去该信号。 第二标量具有比例率,其是第一标量的比例率的倒数,使得在近端路径上发生单位缩放。
    • 4. 发明授权
    • Controlling attenuation during echo suppression
    • 控制回波抑制期间的衰减
    • US07065207B2
    • 2006-06-20
    • US10660446
    • 2003-09-11
    • David L. BarronWilliam C. YipSean S. You
    • David L. BarronWilliam C. YipSean S. You
    • H04B3/23
    • H04M9/082
    • An echo canceling system receives and transmits audio signals between a far end and a near end. During single talk, which is when only one end is originating audio, the path back to the originator is impeded by echo cancellation and attenuation. When there is double talk, which is when both ends are originating audio, the attenuation is removed, or at least significantly reduced. This is achieved by using ERLE, which itself is a known signal used for other purposes in an echo cancellation system, to provide information as to when double talking is occurring. This allows for stopping the attenuation for the double talk situation, which is the desired result.
    • 回波消除系统在远端和近端之间接收和发送音频信号。 在单通话中,当只有一端是起始音频时,回到起始器的路径受到回波消除和衰减的阻碍。 当双方谈话时,当两端是起始音频时,衰减被去除,或至少显着减少。 这是通过使用ERLE来实现的,该ERLE本身是用于回波消除系统中的其它目的的已知信号,以提供关于双重通话发生的信息。 这允许停止双重通话情况的衰减,这是期望的结果。
    • 5. 发明授权
    • Communication system apparatus for transmitting and receiving data
having a radio wireline interface
    • 用于发送和接收具有无线电线接口的数据的通信系统装置
    • US5504802A
    • 1996-04-02
    • US103389
    • 1993-08-09
    • Paul R. KennedyWilliam C. YipTimothy G. Hall
    • Paul R. KennedyWilliam C. YipTimothy G. Hall
    • H04B7/185H04Q7/38
    • H04B7/18532
    • A digital satellite communication system including a local terminal coupled to a radio wireline interface through a radio satellite network. The radio wireline interface connects the radio satellite network to a public switched telephone network and a remote communication terminal. The system includes a novel method of establishing an end-to-end communication channel between local and remote terminals wherein the local terminal establishes a direct digital channel between itself and the radio wireline interface and transmits a message describing its signaling capabilities to the radio wireline interface. The radio wireline interface then trains its modem with the modem of the remote terminal such that the signaling capabilities of the local terminal are not violated. By moving the modem training procedure to the radio wireline interface, modem training response delays caused by the radio satellite network do not affect the success of establishing the end-to-end communication channel.
    • 一种数字卫星通信系统,包括通过无线电卫星网络耦合到无线电有线接口的本地终端。 无线电有线接口将无线电卫星网络连接到公共交换电话网络和远程通信终端。 该系统包括在本地和远程终端之间建立端到端通信信道的新方法,其中本地终端在其本身和无线电有线接口之间建立直接数字信道,并将描述其信令能力的消息发送到无线电有线接口 。 然后,无线电有线接口用远程终端的调制解调器训练其调制解调器,使得本地终端的信令能力不被违反。 通过将调制解调器训练程序移动到无线电有线接口,由无线电卫星网络引起的调制解调器训练响应延迟不影响建立端到端通信信道的成功。
    • 6. 发明授权
    • Method and apparatus for volume switched gain control
    • 用于音量切换增益控制的方法和装置
    • US5357567A
    • 1994-10-18
    • US930254
    • 1992-08-14
    • David L. BarronJames A. StephensWilliam C. Yip
    • David L. BarronJames A. StephensWilliam C. Yip
    • H03G3/20H04M1/60H04M1/00
    • H03G3/3089H04M1/6033
    • An apparatus comprises a first and a second input and a processor coupled to the inputs. The processor estimates peak and minimum levels of each of the first and second input signals. A first signal regulator is coupled to the first input and to a first output for delivering a first output signal from the first signals. A second signal regulator is coupled to the second input and to a second output for delivering a second output signal from the second audio signals. A gain adjustment device is coupled to the processor and the first and second signal regulators. The gain adjustment device provides control signals to the first and second signal regulators to adjust an output signal level of the first output signal in response to minimum and peak levels of the first and second input signals. The output signal level is continuously variable over a range.
    • 一种装置包括第一和第二输入以及耦合到输入的处理器。 处理器估计第一和第二输入信号中的每一个的峰值和最小电平。 第一信号调节器耦合到第一输入端和第一输出端,​​用于从第一信号传送第一输出信号。 第二信号调节器耦合到第二输入端和第二输出端,用于从第二音频信号传送第二输出信号。 增益调节装置耦合到处理器和第一和第二信号调节器。 增益调整装置向第一和第二信号调节器提供控制信号,以响应于第一和第二输入信号的最小和峰值电平来调整第一输出信号的输出信号电平。 输出信号电平在一个范围内连续变化。
    • 8. 发明授权
    • Efficient codebook search for CELP vocoders
    • 高效的码本搜索CELP声码器
    • US5187745A
    • 1993-02-16
    • US722572
    • 1991-06-27
    • William C. YipDavid L. Barron
    • William C. YipDavid L. Barron
    • G10L19/00G10L19/12
    • G10L19/12G10L25/06G10L25/18
    • A new way of determining correlation coefficients for stochastic codebook vectors for CELP coding of speech takes advantage of the sparsely populated nature of stochastic codebook vectors. N valued input signals (e.g., convolution vectors) to be correlated with N valued codebook vectors are fed to an N by N multiplexer or other selection means and the signal values either passed to an accumulator or not according to the state of N select inputs or other identification means determined from a memory store (e.g., an EPROM) whose entries correspond to the non-zero values of the codebook vectors. The accumulator output is the correlation of the codebook vector with the input signal. A sequencer steps through the entire codebook to provide correlation values for each vectors. The results are used to determine the optimum stochastic codebook vector for replicating the particular speech frame being analyzed.
    • 确定语音CELP编码的随机码本向量的相关系数的新方法利用随机码本向量的人口稀疏性质。 与N值码本矢量相关联的N值输入信号(例如,卷积矢量)被馈送到N乘N多路复用器或其他选择装置,并且信号值根据N个选择输入的状态或不是传送到累加器, 从存储器存储(例如,EPROM)确定的其他识别装置,其存储对应于码本矢量的非零值。 累加器输出是码本向量与输入信号的相关。 定序器遍历整个码本,为每个向量提供相关值。 结果用于确定用于复制被分析的特定语音帧的最优随机码本向量。
    • 9. 发明授权
    • Efficient calculation of autocorrelation coefficients for CELP vocoder
adaptive codebook
    • CELP声码器自适应码本的自相关系数的有效计算
    • US5179594A
    • 1993-01-12
    • US714409
    • 1991-06-12
    • William C. YipDavid L. Barron
    • William C. YipDavid L. Barron
    • G10L19/00G10L19/12
    • G10L19/12G10L25/06G10L25/18
    • A new way of determining autocorrelation coefficients for adaptive codebook vectors for CELP coding of speech simplifies and improves the accuracy of the autocorrelation coefficient determination for the situation where the codebook vector length being analyzed is less than a speech frame length. This is important in synthesizing short pitch period speech. Copy-up of the shortened codebook vector to equal the frame length is not needed and autocorrelation coefficient errors associated with copy-up are avoided. The improved system relies on calculating autocorrelation coefficients of the first (shortest) vector and then obtaining subsequent autocorrelation coefficients for successive vectors of increasing length by a simple end correction technique until the vector length equals the frame length. The autocorrelation coefficients are scaled by multiplying them by the ratio of the frame length to the vector length.
    • 确定语音CELP编码的自适应码本向量的自相关系数的新方式简化并提高了自动相关系数确定的准确性,其中所分析的码本矢量长度小于语音帧长度。 这在合成短节距语音中是重要的。 将缩短的码本向量复制为等于帧长度,并避免与复制相关的自相关系数误差。 改进的系统依赖于计算第一(最短)向量的自相关系数,然后通过简单的结束校正技术获得随后的增长长度向量的随机自相关系数,直到向量长度等于帧长度。 通过将自相关系数乘以帧长度与向量长度的比率来缩放。