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    • 3. 发明授权
    • Sensor array beamformer post-processor
    • 传感器阵列波束形成器后处理器
    • US09054764B2
    • 2015-06-09
    • US13187235
    • 2011-07-20
    • Ivan TashevAlejandro Acero
    • Ivan TashevAlejandro Acero
    • H04R3/00H04B7/08
    • H04B7/0854
    • A novel beamforming post-processor technique with enhanced noise suppression capability. The present beamforming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction, resulting in minimal artifacts and musical noise.
    • 一种具有增强噪声抑制能力的新型波束成形后处理器技术。 本波束形成后处理器技术是用于传感器阵列(例如麦克风阵列)的非线性后处理技术,其改善了方向性和信号分离能力。 该技术在所谓的瞬时到达空间方向上工作,估计来自给定入射角或查找方向的声音的概率,并且应用时间变化的基于增益的时空滤波器来抑制来自其他方向的声音 比声源方向,导致最小的伪影和音乐噪音。
    • 6. 发明申请
    • FACTORED TRANSFORMS FOR SEPARABLE ADAPTATION OF ACOUSTIC MODELS
    • 用于可分离适应声学模型的变换
    • US20130253930A1
    • 2013-09-26
    • US13427907
    • 2012-03-23
    • Michael Lewis SeltzerAlejandro Acero
    • Michael Lewis SeltzerAlejandro Acero
    • G10L15/00
    • G10L15/063G10L15/07G10L15/20
    • Various technologies described herein pertain to adapting a speech recognizer to input speech data. A first linear transform can be selected from a first set of linear transforms based on a value of a first variability source corresponding to the input speech data, and a second linear transform can be selected from a second set of linear transforms based on a value of a second variability source corresponding to the input speech data. The linear transforms in the first and second sets can compensate for the first variability source and the second variability source, respectively. Moreover, the first linear transform can be applied to the input speech data to generate intermediate transformed speech data, and the second linear transform can be applied to the intermediate transformed speech data to generate transformed speech data. Further, speech can be recognized based on the transformed speech data to obtain a result.
    • 本文描述的各种技术涉及使语音识别器适应于输入语音数据。 可以基于与输入语音数据相对应的第一可变性源的值从第一组线性变换中选择第一线性变换,并且可以基于第二组线性变换的值,从第二组线性变换中选择第二线性变换 对应于输入语音数据的第二可变性源。 第一和第二组中的线性变换可以分别补偿第一可变性源和第二可变性源。 此外,可以将第一线性变换应用于输入语音数据以产生中间变换语音数据,并且可以将第二线性变换应用于中间变换语音数据以生成变换语音数据。 此外,可以基于变换的语音数据来识别语音以获得结果。
    • 9. 发明授权
    • Phase sensitive model adaptation for noisy speech recognition
    • 嘈杂语音识别的相敏模型适应
    • US08214215B2
    • 2012-07-03
    • US12236530
    • 2008-09-24
    • Jinyu LiLi DengDong YuYifan GongAlejandro Acero
    • Jinyu LiLi DengDong YuYifan GongAlejandro Acero
    • G10L15/14
    • G10L15/065G10L15/20
    • A speech recognition system described herein includes a receiver component that receives a distorted speech utterance. The speech recognition also includes an updater component that is in communication with a first model and a second model, wherein the updater component automatically updates parameters of the second model based at least in part upon joint estimates of additive and convolutive distortions output by the first model, wherein the joint estimates of additive and convolutive distortions are estimates of distortions based on a phase-sensitive model in the speech utterance received by the receiver component. Further, distortions other than additive and convolutive distortions, including other stationary and nonstationary sources, can also be estimated used to update the parameters of the second model.
    • 本文描述的语音识别系统包括接收失真的语音话语的接收机组件。 所述语音识别还包括与第一模型和第二模型通信的更新器组件,其中所述更新器组件至少部分地基于由所述第一模型输出的加法和卷积失真的联合估计来自动更新所述第二模型的参数 其中,加法和卷积失真的联合估计是基于由接收器部件接收的语音发声中的相敏模型的失真估计。 此外,还可以估计用于更新第二模型参数的除加法和卷积失真之外的失真,包括其他静止和非平稳源。
    • 10. 发明授权
    • Piecewise-based variable-parameter Hidden Markov Models and the training thereof
    • 基于分段的可变参数隐马尔科夫模型及其训练
    • US08160878B2
    • 2012-04-17
    • US12211114
    • 2008-09-16
    • Dong YuLi DengYifan GongAlejandro Acero
    • Dong YuLi DengYifan GongAlejandro Acero
    • G10L15/14G10L15/20
    • G10L15/144
    • A speech recognition system uses Gaussian mixture variable-parameter hidden Markov models (VPHMMs) to recognize speech under many different conditions. Each Gaussian mixture component of the VPHMMs is characterized by a mean parameter μ and a variance parameter Σ. Each of these Gaussian parameters varies as a function of at least one environmental conditioning parameter, such as, but not limited to, instantaneous signal-to-noise-ratio (SNR). The way in which a Gaussian parameter varies with the environmental conditioning parameter(s) can be approximated as a piecewise function, such as a cubic spline function. Further, the recognition system formulates the mean parameter μ and the variance parameter Σ of each Gaussian mixture component in an efficient form that accommodates the use of discriminative training and parameter sharing. Parameter sharing is carried out so that the otherwise very large number of parameters in the VPHMMs can be effectively reduced with practically feasible amounts of training data.
    • 语音识别系统使用高斯混合可变参数隐马尔可夫模型(VPHMM)来识别许多不同条件下的语音。 VPHMM的每个高斯混合分量的特征在于平均参数μ和方差参数&Sgr。 这些高斯参数中的每一个作为至少一个环境调节参数的函数而变化,例如但不限于瞬时信噪比(SNR)。 高斯参数随环境条件参数变化的方式可以近似为分段函数,如三次样条函数。 此外,识别系统制定均值参数μ和方差参数&Sgr; 每个高斯混合分量以有效的形式适应使用歧视性训练和参数共享。 执行参数共享,以便通过实际可行的训练数据量可以有效地减少VPHMM中非常大量的参数。