公开(公告)号：US08213635B2

公开(公告)日：2012-07-03

申请号：US12328789

申请日：2008-12-05

Applicant: Qin Li ,  Michael Lewis Seltzer ,  Chao He

Inventor： Qin Li ,  Michael Lewis Seltzer ,  Chao He

IPC: H04B15/00 ,  G10L11/06

CPC classification number: G10L21/0208 ,  G10L21/0232

Abstract: An audio signal is received that might include keyboard noise and speech. The audio signal is digitized and transformed from a time domain to a frequency domain. The transformed audio is analyzed to determine whether there is likelihood that keystroke noise is present. If it is determined there is high likelihood that the audio signal contains keystroke noise, a determination is made as to whether a keyboard event occurred around the time of the likely keystroke noise. If it is determined that a keyboard event occurred around the time of the likely keystroke noise, a determination is made as to whether speech is present in the audio signal around the time of the likely keystroke noise. If no speech is present, the keystroke noise is suppressed in the audio signal. If speech is detected in the audio signal or if the keystroke noise abates, the suppression gain is removed from the audio signal.

Abstract translation: 接收到可能包括键盘噪音和语音的音频信号。 音频信号被数字化并从时域变换到频域。 分析变换的音频以确定是否存在按键噪声的可能性。 如果确定音频信号包含击键噪声的可能性很高，则确定在可能的击键噪声的时间周围是否发生键盘事件。 如果确定在可能的击键噪声的时间周围发生键盘事件，则确定在可能的击键噪声的时间周围的音频信号中是否存在语音。 如果不存在语音，则在音频信号中抑制击键噪声。 如果在音频信号中检测到语音，或者如果击键噪声消失，则从音频信号中去除抑制增益。

公开(公告)号：USD645026S1

公开(公告)日：2011-09-13

申请号：US29353846

申请日：2010-01-14

Applicant: Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J Lamb

Designer： Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J Lamb

公开(公告)号：USD645024S1

公开(公告)日：2011-09-13

申请号：US29378325

申请日：2010-11-02

Applicant: Yves Albert Behar ,  Gabriel J. Lamb ,  Qin Li

Designer： Yves Albert Behar ,  Gabriel J. Lamb ,  Qin Li

公开(公告)号：USD633480S1

公开(公告)日：2011-03-01

申请号：US29354212

申请日：2010-01-20

Applicant: Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J. Lamb

Designer： Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J. Lamb

公开(公告)号：USD632676S1

公开(公告)日：2011-02-15

申请号：US29353847

申请日：2010-01-14

Applicant: Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J Lamb

Designer： Yves Albert Behar ,  Qin Li ,  Noah Murphy-Reinhertz ,  Gabriel J Lamb

公开(公告)号：US07764634B2

公开(公告)日：2010-07-27

申请号：US11275414

申请日：2005-12-29

Applicant: Wei Zhong ,  Chao He ,  Anton W. Krantz ,  Qin Li

Inventor： Wei Zhong ,  Chao He ,  Anton W. Krantz ,  Qin Li

IPC: H04B3/20

CPC classification number: H04M9/08 ,  H04M1/20 ,  H04M1/2535 ,  H04M1/6033 ,  H04M9/082

Abstract: Suppressing one or more frequency ranges of a signal prevents the occurrence of feedback in a voice data communications application. A system recognizes a frequency range in a signal where feedback occurs, or anticipates a frequency range where feedback is anticipated. The signal includes a signal the input system generates or that the output system renders. The system suppresses the signal in the frequency range by disregarding one or more sampling bits representing the frequency range, or by applying one or more filters to attenuate or eliminate the signal in the frequency range. The system may monitor the signal to identify feedback resulting in different or additional frequency ranges and suppress the signal in the different or additional frequency ranges to prevent feedback from occurring.

Abstract translation: 抑制信号的一个或多个频率范围阻止语音数据通信应用中的反馈的发生。 系统识别发生反馈的信号中的频率范围，或者预期预期反馈的频率范围。 信号包括输入系统产生的信号或输出系统呈现的信号。 该系统通过忽略表示频率范围的一个或多个采样位，或通过施加一个或多个滤波器来衰减或消除频率范围内的信号来抑制频率范围内的信号。 该系统可以监视信号以识别导致不同或附加频率范围的反馈，并且抑制不同或附加频率范围内的信号以防止发生反馈。

公开(公告)号：US20080247535A1

公开(公告)日：2008-10-09

申请号：US11784692

申请日：2007-04-09

Applicant: Qin Li ,  Chao He

Inventor： Qin Li ,  Chao He

IPC: H04M9/08

CPC classification number: H04M9/082 ,  H04L1/0025 ,  H04L1/0026

Abstract: A method is provided for reducing the adverse impact of echo on audio quality in a two-communication system. The method includes two parts. The first part begins by detecting non-linear effects (e.g. clippings and audio glitches). If a non-linear effect is detected, the system temporarily disables adaptation of the adaptive filter. In this way, filter coefficients obtained before the non-linear effect happens will not be corrupted, so the AEC can quickly recover from the non-linear effects. The second part begins by monitoring a parameter reflecting signal quality (e.g., ERLE). If the signal quality parameter falls below a given value the system switches from a full-duplex mode of operation to a half-duplex mode of operation. In this way, when a non-linear effect that is undetectable or occurs repeatedly (e.g., speaker volume changes) and which can corrupt an acoustic echo canceller (AEC) for a relatively long period of time, the system switches from full-duplex operation to half-duplex operation. In half-duplex operation, communication can only happen in one direction at a time, and thus the echo path is broken, effectively eliminating echoes. When the non-linear effect is no longer present and the quality parameter rises to a normal level, communication returns to a full-duplex mode of operation and the AEC once again removes the echoes.

Abstract translation: 提供一种用于减少回声对双通信系统中音频质量的不利影响的方法。 该方法包括两部分。 第一部分通过检测非线性效果（例如剪辑和音频毛刺）开始。 如果检测到非线性效应，则系统暂时禁用自适应滤波器的自适应。 以这种方式，在非线性效应发生之前获得的滤波器系数不会被破坏，因此AEC可以从非线性效应快速恢复。 第二部分通过监视反映信号质量的参数（例如，ERLE）开始。 如果信号质量参数低于给定值，系统将从全双工工作模式切换到半双工工作模式。 以这种方式，当一个不可检测的或重复发生的非线性效应（例如，扬声器音量变化）并且可以在相对长的时间段内损坏声学回声消除器（AEC）时，系统从全双工操作切换 进行半双工操作。 在半双工操作中，通信只能一次在一个方向发生，因此回波路径被破坏，有效地消除了回波。 当非线性效应不再存在并且质量参数上升到正常水平时，通信返回到全双工操作模式，并且AEC再次移除回波。

公开(公告)号：US20070263849A1

公开(公告)日：2007-11-15

申请号：US11413710

申请日：2006-04-28

Applicant: Jack Stokes ,  Chao He ,  Qin Li

Inventor： Jack Stokes ,  Chao He ,  Qin Li

IPC: H04M9/08

CPC classification number: H04M9/082

Abstract: Center clipping is applied with acoustic echo suppression in a two-way voice communication system to reduce a microphone signal to the background noise floor when speech is not present. For integration with a microphone array, the center clipping processing determines whether speech is present based on estimates of the overall leak through and instantaneous microphone power across the microphone array channels. The overall estimates can be calculated as a dot product of the microphone array coefficients computed by a sound source localization process and separate estimates for the respective microphone channel.

Abstract translation: 在双向语音通信系统中，中心限幅被应用于声学回声抑制，以在不存在语音时将麦克风信号减小到背景噪声底层。 为了与麦克风阵列集成，基于麦克风阵列通道上的总泄漏和瞬时麦克风功率的估计，中心削波处理确定语音是否存在。 可以将总体估计值计算为由声源定位处理计算的麦克风阵列系数和各个麦克风通​​道的单独估计的点乘积。

公开(公告)号：US20070165838A1

公开(公告)日：2007-07-19

申请号：US11332500

申请日：2006-01-13

Applicant: Qin Li ,  Chao He ,  Wei-Ge Chen ,  Mu Han

Inventor： Qin Li ,  Chao He ,  Wei-Ge Chen ,  Mu Han

IPC: A61F11/06 ,  H04M9/08 ,  G10K11/16 ,  H03B29/00

CPC classification number: H04M9/082

Abstract: The quality and robustness of audio echo cancellation is enhanced by selectively applying glitch recovery processes based on a quality measurement of the relative offset between capture and render audio streams. For example, large and small glitch detection is enabled for low relative offset variance; large glitch detection is enabled in a medium range of relative offset variance; and neither enabled at high variance. Further, a fast glitch recovery process suspends updating the adaptive filter coefficients of the audio echo cancellation while buffers are re-aligned to recover from the glitch, so as to avoid resetting the adaptive filter. When clock drift exists between capture and render audio streams, a multi-step compensation method is applied to improve AEC output quality in case the drifting rate is low; and a resampler is used to compensate the drift in case the drifting rate is high. An anti-clipping process detects clipping of the signals, and also suspends adaptive filter updating during clipping.

Abstract translation: 通过基于捕获和渲染音频流之间的相对偏移的质量测量来选择性地应用毛刺恢复过程来增强音频回声消除的质量和鲁棒性。 例如，对于低相对偏移方差，启用大的和小的毛刺检测; 在相对偏移方差的中等范围内启用大毛刺检测; 并且在高方差时都不启用。 此外，快速毛刺恢复处理暂停更新音频回声消除的自适应滤波器系数，同时缓冲器被重新对准以从毛刺恢复，以避免复位自适应滤波器。 当捕获和渲染音频流之间存在时钟漂移时，应用多步补偿方法来提高漂移率低的AEC输出质量; 并且在漂移速率高的情况下使用重采样器来补偿漂移。 反剪辑过程检测信号的剪辑，并且还可以在剪辑期间暂停自适应滤波器更新。

公开(公告)号：US09450756B2

公开(公告)日：2016-09-20

申请号：US13879619

申请日：2010-12-22

Applicant: Zhiqiang Du ,  Manxia Tie ,  Guoqiang Zhang ,  Qin Li

Inventor： Zhiqiang Du ,  Manxia Tie ,  Guoqiang Zhang ,  Qin Li

IPC: H04L9/32 ,  H04W12/06

CPC classification number: H04L9/32 ,  H04L9/3271 ,  H04W12/06

Abstract: A method and a system for authenticating an entity based on a symmetric encryption algorithm are provided. The method includes the following steps: 1) an entity A sends an authentication request message to an entity B; 2) after receiving the authentication request message, the entity B sends an authentication response message to the entity A; 3) the entity A determines the validity of the entity B according to the received authentication response message. The implementation cost of the system can be reduced by using the authentication according to the invention.

Abstract translation: 提供了一种基于对称加密算法认证实体的方法和系统。 该方法包括以下步骤：1）实体A向实体B发送认证请求消息; 2）接收到认证请求报文后，实体B向实体A发送认证响应消息; 3）实体A根据收到的认证响应消息确定实体B的有效性。 通过使用根据本发明的认证可以减少系统的实现成本。