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1.发明申请Methods, systems, and circuits for text independent speaker recognition with automatic learning features 有权用于具有自动学习功能的文本独立扬声器识别的方法，系统和电路公开(公告)号：US20140195232A1
公开(公告)日：2014-07-10
申请号：US13854134
申请日：2013-04-01
Applicant: STMicroelectronics Asia Pacific Pte Ltd.
Inventor： Evelyn Kurniawati , Sapna George
IPC: G10L17/04
CPC classification number: G10L17/04 , G10L15/063 , G10L15/144 , G10L17/02 , G10L17/06
Abstract: Embodiments provide a method and system of text independent speaker recognition with a complexity comparable to a text dependent version. The scheme exploits the fact that speech is a quasi-stationary signal and simplifies the recognition process based on this theory. The modeling allows the speaker profile to be updated progressively with the new speech sample that is acquired during usage time.
Abstract translation: 实施例提供了具有与文本相关版本相当的复杂性的文本独立说话人识别的方法和系统。 该方案利用了语音是准稳态信号，并简化了基于该理论的识别过程。 该建模允许使用在使用时间期间获取的新的语音样本来逐渐更新扬声器简档。
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2.发明授权Methods, systems, and circuits for text independent speaker recognition with automatic learning features 有权用于具有自动学习功能的文本独立扬声器识别的方法，系统和电路公开(公告)号：US09530417B2
公开(公告)日：2016-12-27
申请号：US13854134
申请日：2013-04-01
Applicant: STMicroelectronics Asia Pacific Pte Ltd.
Inventor： Evelyn Kurniawati , Sapna George
CPC classification number: G10L17/04 , G10L15/063 , G10L15/144 , G10L17/02 , G10L17/06
Abstract: Methods and systems of text independent speaker recognition provide a complexity comparable to text dependent speaker recognition system. These methods and systems exploit the fact that speech is a quasi-stationary signal and simplify the recognition process based on this theory. The speaker modeling allows a speaker profile to be updated progressively with new speech samples that are acquired during usage over time by the speaker.
Abstract translation: 文本独立说话人识别的方法和系统提供了与文本相关的扬声器识别系统相当的复杂性。 这些方法和系统利用语音是准稳态信号，并且基于这个理论简化识别过程。 扬声器建模允许使用由扬声器随着时间的使用获取的新语音样本来逐渐更新扬声器简档。
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3.发明申请ADAPTIVE LOUDNESS LEVELLING METHOD FOR DIGITAL AUDIO SIGNALS IN FREQUENCY DOMAIN 有权频域中数字音频信号的自适应朗读方法公开(公告)号：US20160191007A1
公开(公告)日：2016-06-30
申请号：US14588398
申请日：2014-12-31
Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTD
Inventor： Wei Li , Sapna George
IPC: H03G3/20
Abstract: Embodiments of the present disclosure are directed to techniques for adjusting the amplitude of a digital audio signal in the frequency domain to control the perceived loudness of the audio signal at a desired level. In one embodiment, a method first adjusts the audio signal to a desired loudness level by applying an adaptive wideband gain and thereafter a multi-band compression is applied to further reduce a dynamic range of the audio signal, and noise analysis and temporal masking operations are also performed to provide a pleasant sound for a listener or listeners.
Abstract translation: 本公开的实施例涉及用于调整频域中的数字音频信号的幅度的技术，以将音频信号的感知响度控制在期望的水平。 在一个实施例中，一种方法首先通过应用自适应宽带增益来将音频信号调整到期望的响度级，然后施加多频带压缩以进一步减小音频信号的动态范围，并且噪声分析和时间屏蔽操作是 还为听众或听众提供了愉快的声音。
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4.发明授权Converting samples of a signal at a sample rate into samples of another signal at another sample rate 有权将采样速率的信号样本转换为另一个采样率的另一个信号的样本公开(公告)号：US08874175B2
公开(公告)日：2014-10-28
申请号：US13672326
申请日：2012-11-08
Applicant: STMicroelectronics Asia Pacific Pte Ltd.
Inventor： Karthik Muralidhar , Sapna George , Saurav Sahu , Frank Teo
Abstract: In an embodiment, an apparatus includes a determiner, converter, adapter, and modifier. The determiner is configured to generate a representation of a difference between a first frequency at which a first signal is sampled and a second frequency at which a second signal is sampled, and the converter is configured to generate a second sample of the first signal at a second time in response to the representation and a first sample of the first signal at a first time. The adapter is configured to generate a sample of a modifier signal in response to the second sample of the first signal, and the modifier is configured to generate a modified sample of the second signal in response to a sample of the second signal and the sample of the modifier signal. For example, such an apparatus may be able to reduce the magnitude of an echo signal in a system having an audio pickup (e.g., a microphone) near an audio output (e.g., a speaker).
Abstract translation: 在一个实施例中，装置包括确定器，转换器，适配器和修改器。 确定器被配置为产生第一信号被采样的第一频率与第二信号被采样的第二频率之间的差异的表示，并且转换器被配置为在第一频率处产生第一信号的第二采样 第二次响应于第一次的表示和第一个信号的第一个样本。 适配器被配置为响应于第一信号的第二样本产生修改器信号的样本，并且修改器被配置为响应于第二信号的样本和第二信号的样本而生成第二信号的修改样本 修改信号。 例如，这样的装置可能能够减小在音频输出（例如扬声器）附近具有音频拾取器（例如，麦克风）的系统中的回波信号的幅度。
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公开(公告)号：US09647624B2
公开(公告)日：2017-05-09
申请号：US14588398
申请日：2014-12-31
Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTD
Inventor： Wei Li , Sapna George
Abstract: Embodiments of the present disclosure are directed to techniques for adjusting the amplitude of a digital audio signal in the frequency domain to control the perceived loudness of the audio signal at a desired level. In one embodiment, a method first adjusts the audio signal to a desired loudness level by applying an adaptive wideband gain and thereafter a multi-band compression is applied to further reduce a dynamic range of the audio signal, and noise analysis and temporal masking operations are also performed to provide a pleasant sound for a listener or listeners.
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Patent Agency Ranking
6.发明授权公开(公告)号：US09633652B2
公开(公告)日：2017-04-25
申请号：US13854133
申请日：2013-03-31
Applicant: STMicroelectronics Asia Pacific Pte Ltd.
Inventor： Evelyn Kurniawati , Sapna George
CPC classification number: G10L15/144 , G10L15/00 , G10L15/06 , G10L15/07 , G10L17/04
Abstract: Embodiments reduce the complexity of speaker dependent speech recognition systems and methods by representing the code phrase (i.e., the word or words to be recognized) using a single Gaussian Mixture Model (GMM) which is adapted from a Universal Background Model (UBM). Only the parameters of the GMM need to be stored. Further reduction in computation is achieved by only checking the GMM component that is relevant to the keyword template. In this scheme, keyword template is represented by a sequence of the index of best performing component of the GMM of the keyword model. Only one template is saved by combining the registration template using Longest Common Sequence algorithm. The quality of the word model is continuously updated by performing expectation maximization iteration using the test word which is accepted as keyword model.
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7.发明申请STEERING VECTOR ESTIMATION FOR MINIMUM VARIANCE DISTORTIONLESS RESPONSE (MVDR) BEAMFORMING CIRCUITS, SYSTEMS, AND METHODS 有权用于最小变化失真响应（MVDR）波导形成电路，系统和方法的转向矢量估计公开(公告)号：US20160192068A1
公开(公告)日：2016-06-30
申请号：US14588288
申请日：2014-12-31
Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTD
Inventor： Samuel Samsudin Ng , Sapna George , Karthik Muralidhar
IPC: H04R1/40
CPC classification number: H04R1/406 , H04R3/005 , H04R2201/40 , H04R2201/401 , H04R2201/403 , H04R2430/23 , H04R2430/25 , H04R2499/11 , H04R2499/13
Abstract: A method of estimating a steering vector of a sensor array of M sensors according to one embodiment of the present disclosure includes estimating a steering vector of a noise source located at an angle 0 degrees from a look direction of the array using a least squares estimate of the gains of the sensors in the array, defining a steering vector of a desired sound source in the look direction of the array, and estimating the steering vector by performing element-by-element multiplication of the estimated noise vector and the complex conjugate of steering vector of the desired sound source. The sensors may be microphones.
Abstract translation: 根据本公开的一个实施例，估计M个传感器的传感器阵列的导向矢量的方法包括使用最小二乘估计来估计与阵列的外观方向成0度角的噪声源的导向矢量 阵列中的传感器的增益，在阵列的外观方向上定义所需声源的导向矢量，以及通过执行所估计的噪声向量和转向的复共轭的逐个元素相乘来估计导向矢量 所需声源的矢量。 传感器可以是麦克风。
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8.发明申请METHODS, SYSTEMS, AND CIRCUITS FOR SPEAKER DEPENDENT VOICE RECOGNITION WITH A SINGLE LEXICON 有权扬声器依赖语音识别的方法，系统和电路与单个LEXICON公开(公告)号：US20140200890A1
公开(公告)日：2014-07-17
申请号：US13854133
申请日：2013-03-31
Applicant: STMicroelectronics Asia Pacific Pte Ltd.
Inventor： Evelyn Kurniawati , Sapna George
IPC: G10L15/14
CPC classification number: G10L15/144 , G10L15/00 , G10L15/06 , G10L15/07 , G10L17/04
Abstract: Embodiments reduce the complexity of speaker dependent speech recognition systems and methods by representing the code word (i.e., the word to be recognized) using a single Gaussian Mixture Model (GMM) which is adapted from a Universal Background Model (UBM). Only the parameters of the GMM need to be stored. Further reduction in computation is achieved by only checking the GMM component that is relevant to the keyword template. In this scheme, keyword template is represented by a sequence of the index of best performing component of the GMM of the keyword model. Only one template is saved by combining the registration template using Longest Common Sequence algorithm. The quality of the word model is continuously updated by performing expectation maximization iteration using the test word which is accepted as keyword model.
Abstract translation: 实施例通过使用从通用背景模型（UBM）改编的单个高斯混合模型（GMM）来表示代码字（即，要识别的单词）来降低说话者依赖语音识别系统和方法的复杂性。 只需要存储GMM的参数。 仅通过检查与关键字模板相关的GMM组件来实现计算的进一步减少。 在该方案中，关键字模板由关键字模型的GMM的最佳执行组件的索引的序列表示。 通过使用最长公共序列算法组合注册模板，仅保存一个模板。 通过使用被接受为关键字模型的测试词来执行期望最大化迭代，不断更新单词模型的质量。
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9.发明授权Steering vector estimation for minimum variance distortionless response (MVDR) beamforming circuits, systems, and methods 有权最小方差失真响应（MVDR）波束成形电路，系统和方法的转向矢量估计公开(公告)号：US09525934B2
公开(公告)日：2016-12-20
申请号：US14588288
申请日：2014-12-31
Applicant: STMICROELECTRONICS ASIA PACIFIC PTE LTD
Inventor： Samuel Samsudin Ng , Sapna George , Karthik Muralidhar
IPC: H04R1/40
CPC classification number: H04R1/406 , H04R3/005 , H04R2201/40 , H04R2201/401 , H04R2201/403 , H04R2430/23 , H04R2430/25 , H04R2499/11 , H04R2499/13
Abstract: A method of estimating a steering vector of a sensor array of M sensors according to one embodiment of the present disclosure includes estimating a steering vector of a noise source located at an angle θ degrees from a look direction of the array using a least squares estimate of the gains of the sensors in the array, defining a steering vector of a desired sound source in the look direction of the array, and estimating the steering vector by performing element-by-element multiplication of the estimated noise vector and the complex conjugate of steering vector of the desired sound source. The sensors may be microphones.
Abstract translation: 根据本公开的一个实施例，估计M个传感器的传感器阵列的导向矢量的方法包括使用最小二乘估计估计与阵列的外观方向成角度θ度的噪声源的导向矢量， 阵列中的传感器的增益，在阵列的外观方向上定义所需声源的导向矢量，以及通过执行所估计的噪声向量和转向的复共轭的逐个元素相乘来估计导向矢量 所需声源的矢量。 传感器可以是麦克风。
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公开(公告)号：US09466304B2
公开(公告)日：2016-10-11
申请号：US14666109
申请日：2015-03-23
Inventor： Peng Zhang , Shuzheng Xu , Pengjun Wang , Sapna George , Huazhong Yang
IPC: G10L19/018
CPC classification number: G10L19/018
Abstract: The present invention is a system and method for digital watermarking, which discloses a system for digital watermarking, to add a watermark to an audio signal generated by a signal source. The system comprises: a spectrum modulator configured to perform spectrum modulation to a watermark bit and a pseudo noise signal to be embedded into the audio signal to generate a modulated signal; a distortion controller coupled to the signal source and the spectrum modulator and configured to shape the modulated signal based on the audio signal, so as to generate a shaped signal satisfying a predetermined distortion constraint; and an interference compensator coupled to the signal source and the distortion controller and configured to generate a compensation signal based on the audio signal, the pseudo noise signal, and the shaped signal, wherein the compensation signal is for compensating for interference to watermark decoding caused by the audio signal.
Abstract translation: 本发明是一种用于数字水印的系统和方法，其公开了一种用于数字水印的系统，用于向由信号源产生的音频信号添加水印。 该系统包括：频谱调制器，被配置为对水印位执行频谱调制，并将伪噪声信号嵌入到音频信号中以产生调制信号; 耦合到所述信号源和所述频谱调制器并且被配置为基于所述音频信号对所述调制信号进行调整的失真控制器，以便产生满足预定失真约束的成形信号; 以及干扰补偿器，其耦合到所述信号源和所述失真控制器，并且被配置为基于所述音频信号，所述伪噪声信号和所述成形信号产生补偿信号，其中所述补偿信号用于补偿由水印解码引起的水印解码的干扰 音频信号。
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