公开(公告)号：US08577677B2

公开(公告)日：2013-11-05

申请号：US12460473

申请日：2009-07-20

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

IPC: G10L21/02

CPC classification number: G10L21/0272 ,  G10L2021/02166

Abstract: A system and method for sound source separation. The system and method use a beamforming technique. The sound source separation system includes a windowing processor; a DFT transformer; a transfer function estimator; and a noise estimator. The system also includes a voice signal extractor that cancels individual voice signals, except an individual voice signal that is desired to be extracted among individual voice signals, from the integrated voice signals. The system further includes a voice signal detector that cancels a noise part provided through the noise estimator from a transfer function of an individual voice signal which is desired to be detected and extracts a noise-canceled individual voice signal. Even when two or more sound sources are simultaneously input, the sound sources can be separated from each other and separately stored and managed, or an initial sound source can be stored and managed.

Abstract translation: 一种声源分离的系统和方法。 该系统和方法使用波束成形技术。 声源分离系统包括开窗处理器; DFT变压器; 传递函数估计器; 和噪声估计器。 该系统还包括语音信号提取器，除了从各个语音信号中提取的单个语音信号之外，还可以从集成语音信号中消除各个语音信号。 该系统还包括语音信号检测器，该语音信号检测器根据期望被检测的单独语音信号的传递函数来消除通过噪声估计器提供的噪声部分，并提取噪声消除的个体语音信号。 即使当同时输入两个或更多个声源时，声源可以彼此分开并分开存储和管理，或者可以存储和管理初始声源。

公开(公告)号：US08774952B2

公开(公告)日：2014-07-08

申请号：US12455873

申请日：2009-06-09

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

IPC: G06F17/00

CPC classification number: G10K11/178 ,  G10K2210/111

Abstract: An adaptive mode control apparatus and method for adaptive beamforming based on detection of a user direction sound are provided. The adaptive mode control apparatus includes a signal intensity detector that searches for signal intensity of each designated direction to detect signal intensity having a maximum value when a voice signal of each direction is input through at least one microphone; and an adaptive mode controller that compares the signal intensity having the maximum value detected through the signal intensity detector with a threshold value and determines whether to perform an adaptive mode of a Generalized Sidelobe Canceller (GSC) according to the comparison results. Therefore, a lack of control of adaptation of an adaptive filter of the conventional art is solved. That is, as one condition for guaranteeing performance of adaptive beamforming, adaptation of an adaptive filter is not performed when noise of a sound with a high autocorrelation is cancelled.

Abstract translation: 提供了一种基于用户方向声音检测的自适应波束成形的自适应模式控制装置和方法。 自适应模式控制装置包括信号强度检测器，当通过至少一个麦克风输入每个方向的语音信号时，搜索每个指定方向的信号强度以检测具有最大值的信号强度; 以及自适应模式控制器，其将具有通过信号强度检测器检测到的最大值的信号强度与阈值进行比较，并且根据比较结果确定是否执行广义旁瓣消除器（GSC）的自适应模式。 因此，解决了传统技术的自适应滤波器的适应性缺乏控制。 也就是说，作为用于保证自适应波束形成的性能的一个条件，当取消具有高自相关性的声音的噪声时，不执行自适应滤波器的自适应。

公开(公告)号：US20100017206A1

公开(公告)日：2010-01-21

申请号：US12460473

申请日：2009-07-20

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Jounghoon Beh ,  Taekjin Lee

IPC: G10L15/20

CPC classification number: G10L21/0272 ,  G10L2021/02166

Abstract: A system and method for sound source separation. The system and method use a beamforming technique. The sound source separation system includes a windowing processor; a DFT transformer; a transfer function estimator; and a noise estimator. The system also includes a voice signal extractor that cancels individual voice signals, except an individual voice signal that is desired to be extracted among individual voice signals, from the integrated voice signals. The system further includes a voice signal detector that cancels a noise part provided through the noise estimator from a transfer function of an individual voice signal which is desired to be detected and extracts a noise-canceled individual voice signal. Even when two or more sound sources are simultaneously input, the sound sources can be separated from each other and separately stored and managed, or an initial sound source can be stored and managed.

Abstract translation: 一种声源分离的系统和方法。 该系统和方法使用波束成形技术。 声源分离系统包括开窗处理器; DFT变压器; 传递函数估计器; 和噪声估计器。 该系统还包括语音信号提取器，除了从各个语音信号中提取的单个语音信号之外，还可以从集成语音信号中消除各个语音信号。 该系统还包括语音信号检测器，该语音信号检测器根据期望被检测的单独语音信号的传递函数来消除通过噪声估计器提供的噪声部分，并提取噪声消除的个体语音信号。 即使当同时输入两个或更多个声源时，声源可以彼此分开并分开存储和管理，或者可以存储和管理初始声源。

公开(公告)号：US20080187152A1

公开(公告)日：2008-08-07

申请号：US12013875

申请日：2008-01-14

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

IPC: H04R3/00

CPC classification number: H04R3/005 ,  G10L21/0208 ,  G10L2021/02166 ,  H04R1/406 ,  H04R2430/20

Abstract: Disclosed are an apparatus and a method for beamforming in consideration of characteristics of an actual noise environment. The apparatus includes a microphone array having at least microphone, the microphone array outputting a signal input through the microphone; a coherence function generation unit for calculating coherences for input signals according to each space between microphones, calculating averages of the coherences for the same distance, and filtering the calculated averages of the coherences and outputting the resultant values, when an input signal is input; a spatial filter factor calculation unit for calculating and outputting a spatial filter factor by using the filtered average coherences; and a beamforming execution unit for performing a beamforming for the input signals by using the spatial filter factor, thereby outputting a noise-processed signal.

Abstract translation: 公开了考虑到实际噪声环境的特性的波束成形的装置和方法。 该装置包括至少具有麦克风的麦克风阵列，麦克风阵列输出通过麦克风输入的信号; 相干函数产生单元，用于根据麦克风之间的每个空间计算输入信号的相干度，计算相同距离的相干的平均值，以及当输入输入信号时滤波所计算的相干的平均值并输出结果值; 空间滤波器因子计算单元，用于通过使用滤波的平均相位来计算和输出空间滤波因子; 以及波束形成执行单元，用于通过使用空间滤波器因子对输入信号执行波束成形，由此输出经噪声处理的信号。

公开(公告)号：US08116478B2

公开(公告)日：2012-02-14

申请号：US12013875

申请日：2008-01-14

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

IPC: H04R3/00

CPC classification number: H04R3/005 ,  G10L21/0208 ,  G10L2021/02166 ,  H04R1/406 ,  H04R2430/20

Abstract: Disclosed are an apparatus and a method for beamforming in consideration of characteristics of an actual noise environment. The apparatus includes a microphone array having at least microphone, the microphone array outputting a signal input through the microphone; a coherence function generation unit for calculating coherences for input signals according to each space between microphones, calculating averages of the coherences for the same distance, and filtering the calculated averages of the coherences and outputting the resultant values, when an input signal is input; a spatial filter factor calculation unit for calculating and outputting a spatial filter factor by using the filtered average coherences; and a beamforming execution unit for performing a beamforming for the input signals by using the spatial filter factor, thereby outputting a noise-processed signal.

Abstract translation: 公开了考虑到实际噪声环境的特性的波束成形的装置和方法。 该装置包括至少具有麦克风的麦克风阵列，麦克风阵列输出通过麦克风输入的信号; 相干函数产生单元，用于根据麦克风之间的每个空间计算输入信号的相干度，计算相同距离的相干的平均值，以及当输入输入信号时滤波所计算的相干的平均值并输出结果值; 空间滤波器因子计算单元，用于通过使用滤波的平均相位来计算和输出空间滤波因子; 以及波束形成执行单元，用于通过使用空间滤波器因子对输入信号执行波束成形，由此输出经噪声处理的信号。

公开(公告)号：US08135586B2

公开(公告)日：2012-03-13

申请号：US12053144

申请日：2008-03-21

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

IPC: G10L19/10

CPC classification number: G10L21/0208

Abstract: Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.

Abstract translation: 公开了一种用于在声音信号处理期间估计包括在声音信号中的噪声的方法和装置。 该方法包括估计输入声音信号的帧中的谐波分量; 使用估计的谐波分量，在输入声音信号的帧上计算声音概率（VPP）; 根据所计算的VPP确定估计噪声谱所必需的方程的权重; 并且使用所确定的权重和所需的等式来估计噪声谱，估计噪声谱和更新噪声谱。

公开(公告)号：US20080235013A1

公开(公告)日：2008-09-25

申请号：US12053144

申请日：2008-03-21

Applicant: Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

Inventor： Hyun-Soo Kim ,  Hanseok Ko ,  Sung-Joo Ahn ,  Jounghoon Beh ,  Hyun-Jin Yoon

IPC: G10L15/20

CPC classification number: G10L21/0208

Abstract: Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.

Abstract translation: 公开了一种用于在声音信号处理期间估计包括在声音信号中的噪声的方法和装置。 该方法包括估计输入声音信号的帧中的谐波分量; 使用估计的谐波分量，在输入声音信号的帧上计算声音概率（VPP）; 根据所计算的VPP确定估计噪声谱所必需的方程的权重; 并且使用所确定的权重和所需的等式来估计噪声谱，估计噪声谱和更新噪声谱。

公开(公告)号：US08849662B2

公开(公告)日：2014-09-30

申请号：US11646911

申请日：2006-12-28

Applicant: Hyun-Soo Kim

Inventor： Hyun-Soo Kim

IPC: G10L15/02 ,  G10L15/00 ,  G10L15/06 ,  G10L17/00 ,  G10L15/04

CPC classification number: G10L15/04 ,  G10L2015/025

Abstract: A method and a system for segmenting phonemes from voice signals. A method for accurately segmenting phonemes, in which a histogram showing a peak distribution corresponding to an order is formed by using a high order concept, and a boundary indicating a starting point and an ending point of each phoneme is determined by calculating a peak statistic based on the histogram. The phoneme segmentation method can remarkably reduce an amount of calculation, and has an advantage of being applied to sound signal systems which perform sound coding, sound recognition, sound synthesizing, sound reinforcement, etc.

Abstract translation: 一种从语音信号中分割音素的方法和系统。 一种用于准确分割音素的方法，其中通过使用高阶概念形成表示与顺序相对应的峰值分布的直方图，并且通过计算基于峰值统计量来确定表示每个音素的起点和终点的边界 在直方图上。 音素分割方法可以显着减少计算量，并且具有应用于执行声音编码，声音识别，声音合成，扩音等的声音信号系统的优点。

公开(公告)号：US08331690B2

公开(公告)日：2012-12-11

申请号：US12726762

申请日：2010-03-18

Applicant: Sang-Wook Oh ,  Seong-Taek Hwang ,  Hyun-Soo Kim ,  Sang-Ho Kim ,  Ho-Seok An ,  Mu-Sik Kwon ,  Ji-Hoon Kim ,  Dong-Chang Lee

Inventor： Sang-Wook Oh ,  Seong-Taek Hwang ,  Hyun-Soo Kim ,  Sang-Ho Kim ,  Ho-Seok An ,  Mu-Sik Kwon ,  Ji-Hoon Kim ,  Dong-Chang Lee

IPC: G06K9/48 ,  G06K9/36 ,  G06K9/32

CPC classification number: G06K9/32 ,  G06K2009/2045 ,  H04N1/3876 ,  H04N5/232 ,  H04N5/23238

Abstract: A method is provided for creating a panorama. The method includes photographing a plurality of images having same backgrounds and different forms of a subject, determining a size and a position of a reference region for creating a panorama using the images, extracting a target region within the reference region from each of the images, detecting same portions in adjacent target regions, and creating a panorama by combining the adjacent target regions on the basis of the same portions.

Abstract translation: 提供了一种用于创建全景图的方法。 该方法包括拍摄具有相同背景和不同形式的对象的多个图像，使用该图像确定用于创建全景的参考区域的大小和位置，从每个图像提取参考区域内的目标区域， 检测相邻目标区域中的相同部分，以及通过基于相同部分组合相邻目标区域来创建全景图。

公开(公告)号：US08135220B2

公开(公告)日：2012-03-13

申请号：US12115250

申请日：2008-05-05

Applicant: Hyun-Soo Kim ,  Je-Han Yoon ,  Hyun-Sik Shim ,  Young-Hee Park ,  Do-Hyung Kim ,  Woo-Han Yun

Inventor： Hyun-Soo Kim ,  Je-Han Yoon ,  Hyun-Sik Shim ,  Young-Hee Park ,  Do-Hyung Kim ,  Woo-Han Yun

IPC: G06K9/62 ,  G06K9/00

CPC classification number: G06K9/00295

Abstract: A face recognition system based on adaptive learning includes a specific person detection and tracking unit for detecting and tracking a specific person from a moving image. A facial feature extraction unit extracts a plurality of facial feature vectors from the detected and tracked specific person. A face recognition unit searches for a given registration model by comparing the extracted facial feature vectors with facial feature vectors of the registration models previously stored in a user registration model database. A learning target selection unit selects a facial feature vector to be added to a record of the given registration model from among the extracted facial feature vectors. A registration model learning unit adds and updates the selected facial feature vector to the record of the given registration model.

Abstract translation: 基于自适应学习的面部识别系统包括用于从运动图像中检测和跟踪特定人物的特定人物检测和跟踪单元。 面部特征提取单元从检测到和跟踪的特定人物中提取多个面部特征向量。 面部识别单元通过将提取的面部特征向量与先前存储在用户登记模型数据库中的登记模型的面部特征向量进行比较来搜索给定的登记模型。 学习目标选择单元从所提取的面部特征向量中选择要添加到给定登记模型的记录的面部特征向量。 注册模型学习单元将所选择的面部特征向量添加并更新到给定注册模型的记录。