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    • 1. 发明申请
    • Method and System for Modeling External Volume Changes Within an Acoustic Echo Canceller
    • 在声学回声消除器内建模外部音量变化的方法和系统
    • US20120250872A1
    • 2012-10-04
    • US13076691
    • 2011-03-31
    • Wilfrid Paul LeBlancElias Nemer
    • Wilfrid Paul LeBlancElias Nemer
    • G10K11/16
    • H04M9/082H04R3/02H04R2420/07H04R2430/01H04R2430/03H04R2499/13
    • A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.
    • 提供了一种方法和系统,其中诸如声学回声消除器的装置可以在外部扬声器音量改变时减少可能在对话的远端听到的残余回声。 设备可以基于由滤波器产生的回波估计和包括音频信息的近端信号来计算增益。 增益可以基于跟踪体积变化的回波估计和近端信号的相关性。 一旦计算,可以验证增益以确保在适当时适用该增益。 可以通过首先将增益应用于滤波器的输出并随后基于增益来缩放滤波器的每个系数的值来调整回波估计。 可以基于若干适配方案在连续帧上平滑增益。
    • 2. 发明授权
    • Method and system for modeling external volume changes within an acoustic echo canceller
    • 用于对声学回声消除器内的外部音量变化进行建模的方法和系统
    • US08644522B2
    • 2014-02-04
    • US13076691
    • 2011-03-31
    • Wilfrid Paul LeBlancElias Nemer
    • Wilfrid Paul LeBlancElias Nemer
    • A61F11/06G10K11/16H03B29/00H04B15/00
    • H04M9/082H04R3/02H04R2420/07H04R2430/01H04R2430/03H04R2499/13
    • A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.
    • 提供了一种方法和系统,其中诸如声学回声消除器的装置可以在外部扬声器音量改变时减少可能在对话的远端听到的残余回声。 设备可以基于由滤波器产生的回波估计和包括音频信息的近端信号来计算增益。 增益可以基于跟踪体积变化的回波估计和近端信号的相关性。 一旦计算,可以验证增益以确保在适当时适用该增益。 可以通过首先将增益应用于滤波器的输出并随后基于增益来缩放滤波器的每个系数的值来调整回波估计。 可以基于若干适配方案在连续帧上平滑增益。
    • 4. 发明授权
    • Single microphone wind noise suppression
    • 单麦克风风噪声抑制
    • US08515097B2
    • 2013-08-20
    • US12261868
    • 2008-10-30
    • Elias NemerWilfrid LeBlancMohammad Zad-IssaJes Thyssen
    • Elias NemerWilfrid LeBlancMohammad Zad-IssaJes Thyssen
    • H04B15/00
    • H04R3/00
    • A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.
    • 描述了用于抑制音频信号中的诸如风噪声之类的非平稳噪声的技术。 根据该技术,分析音频信号的一系列帧以检测音频信号是否包括非平稳噪声。 如果检测到音频信号包括非平稳噪声,则执行多个步骤。 根据这些步骤,确定音频信号的帧是否包括非平稳噪声或语音和非平稳噪声。 如果确定帧包括非平稳噪声,则将第一滤波器应用于帧,并且如果确定帧包括语音和非平稳噪声,则将第二滤波器应用于帧。
    • 7. 发明申请
    • SPEAKER LOCALIZATION SYSTEM AND METHOD
    • 扬声器本地化系统和方法
    • US20100217590A1
    • 2010-08-26
    • US12391879
    • 2009-02-24
    • Elias NemerJes Thyssen
    • Elias NemerJes Thyssen
    • G10L15/20
    • G01S3/86G01S3/8006G10L15/26G10L2021/02166
    • A system and method for performing speaker localization is described. The system and method utilizes speaker recognition to provide an estimate of the direction of arrival (DOA) of speech sound waves emanating from a desired speaker with respect to a microphone array included in the system. Candidate DOA estimates may be preselected or generated by one or more other DOA estimation techniques. The system and method is suited to support steerable beamforming as well as other applications that utilize or benefit from DOA estimation. The system and method provides robust performance even in systems and devices having small microphone arrays and thus may advantageously be implemented to steer a beamformer in a cellular telephone or other mobile telephony terminal featuring a speakerphone mode.
    • 描述用于执行扬声器定位的系统和方法。 系统和方法利用说话者识别来提供相对于包括在系统中的麦克风阵列从期望的扬声器发出的语音声波的到达方向(DOA)的估计。 候选DOA估计可以由一个或多个其它DOA估计技术预先选择或产生。 该系统和方法适用于支持可导向波束形成以及利用或受益于DOA估计的其他应用。 该系统和方法即使在具有小麦克风阵列的系统和设备中也提供了强大的性能,因此可以有利地实现以引导具有扬声器电话模式的蜂窝电话或其他移动电话终端中的波束形成器。