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    • 14. 发明授权
    • Correlation-based two microphone algorithm for noise reduction in reverberation
    • 基于相关性的两个麦克风算法,用于混响中降噪
    • US09489963B2
    • 2016-11-08
    • US14658873
    • 2015-03-16
    • QUALCOMM Technologies International, Ltd.
    • Nima Yousefian JaziRogerio Guedes Alves
    • G10L21/0208G10L21/0316G10L21/0216
    • G10L21/0208G10L21/0232G10L21/0264G10L21/0316G10L2021/02165G10L2021/02166
    • Embodiments are directed towards providing speech enhancement of audio signals from a target source and noise reduction of audio signals from a noise source. A coherence between a first audio signal from a first microphone and a second audio signal from a second microphone may be determined. A first gain function may be determined based on real components of a coherence function, wherein the real components include coefficients based on the previously determined coherence. A second gain function may be determined based on imaginary components of the coherence function. And a third gain function may be determined based on a relationship between a real component of the coherence function and a threshold range. An enhanced audio signal may be generated by applying a combination of the first gain function, the second gain function, and the third gain function to the first audio signal.
    • 实施例旨在提供来自目标源的音频信号的语音增强和来自噪声源的音频信号的降噪。 可以确定来自第一麦克风的第一音频信号和来自第二麦克风的第二音频信号之间的相干性。 可以基于相干函数的实际分量来确定第一增益函数,其中,实际分量包括基于先前确定的相干性的系数。 可以基于相干函数的虚部来确定第二增益函数。 并且可以基于相干函数的实部分量与阈值范围之间的关系来确定第三增益函数。 可以通过将第一增益函数,第二增益函数和第三增益函数的组合应用于第一音频信号来生成增强音频信号。
    • 16. 发明授权
    • Method and system for bias corrected speech level determination
    • 用于偏差校正语音级别确定的方法和系统
    • US09373341B2
    • 2016-06-21
    • US14384586
    • 2013-03-21
    • DOLBY LABORATORIES LICENSING CORPORATION
    • David GunawanGlenn Dickins
    • G10L25/78G10L21/0316G10L25/48G10L25/18G10L25/21
    • G10L21/0316G10L25/18G10L25/21G10L25/48G10L25/78
    • Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.
    • 一种用音频信号测定的语音水平的方法,该方法通过增加噪声对其进行修正和/或降低其变化的影响和/或对其进行幅度压缩,以及被配置为执行该方法的任何实施例的系统 。 在一些实施例中,该方法包括以下步骤:产生表示输入语音信号的频带,频域数据,根据数据确定语音信号的高斯参数频谱模型,以及从参数频谱模型确定估计的平均语音电平 和数据的每个频带的标准偏差值; 以及生成指示针对每个频带的偏置校正的平均语音电平的语音电平数据,包括使用至少一个校正值来校正所述频带的估计平均语音电平,其中每个校正值已经使用参考语音模型预先确定。
    • 19. 发明申请
    • AUDIO NOISE REDUCTION CIRCUIT, AN INTELLIGENT TERMINAL USING THIS AUDIO NOISE REDUCTION CIRCUIT AS WELL AS A TEACHING METHOD
    • 音频噪声减少电路,使用该音频噪声减少电路的智能终端,作为教学方法
    • US20160111009A1
    • 2016-04-21
    • US14667337
    • 2015-03-24
    • SHENZHEN SHISHANGDEYUAN CULTURAL DIFFUSION CO., LTD.
    • QIWEI LU
    • G09B5/06G10L21/0316G10K11/16
    • G09B5/065G10K11/16G10L21/0208G10L21/0272G10L21/0316H04R3/005H04R2410/05
    • The invention provides an audio noise reduction circuit, an intelligent terminal and a teaching method using this audio noise reduction circuit, where a dual microphone array is used for abatement of noise, ICA algorithm is used for blind source analysis; the audio noise reduction circuit is used to realize the extraction and separation of voice and improve the voice quality in noisy environment, so that the user can also obtain clear voice communication or recording effect even in noise environment; through real-time acquisition and recording of the audio information of teacher, the invention realizes the real-time recording of teaching voice data and solves the following problem: when the learning content is preset in the intelligent terminal, the learning experience is relatively monotonous, leading to unsatisfactory results of learning. Additionally, the invention also realizes the wireless sharing of high quality teachers and the real-time synchronous feedback in the process of teaching and learning, featuring such advantages as high interaction and good results of learning.
    • 本发明提供了一种使用该音频降噪电路的音频降噪电路,智能终端和教学方法,其中双麦克风阵列用于消除噪声,ICA算法用于盲源分析; 音频降噪电路用于实现语音提取和分离,提高嘈杂环境中的语音质量,使用户即使在噪声环境下也可以获得清晰的语音通信或记录效果; 通过实时采集和记录教师的音频信息,本发明实现了语音数据教学的实时记录,解决了以下问题:当学习内容在智能终端中预设时,学习体验相对单调, 导致学习效果不理想。 此外,本发明还实现了高质量教师的无线共享和教学过程中的实时同步反馈,具有交互性高,学习效果好等优点。