다채널 잡음처리 장치 및 방법
    31.
    发明授权
    다채널 잡음처리 장치 및 방법 有权
    多通道噪声减少的方法和装置

    公开(公告)号:KR101082839B1

    公开(公告)日:2011-11-11

    申请号:KR1020080131238

    申请日:2008-12-22

    Abstract: 본발명은다채널잡음처리장치및 방법에관한것으로, 다채널음성인식환경에기반하여다채널잡음처리방식가운데빔포밍방식과음원분리방식을잡음처리성능이최대가되도록환경조건에따라선택하여적용함으로써음성인식의성능을향상시킬수 있으며, 특히, 로봇에음성인식을적용하는환경에서는고정된형태및 위치가아닌다양한형태및 위치의잡음원이존재할수 있으므로음성인식을수행할상황에적합한잡음처리가적용되어야하기에, 본발명에의하면효과적인잡음처리를제공하여음성인식의성능을향상시킬수 있는이점이있다.

    음성신호에서 통계적 모델을 이용한 잡음 제거 장치 및 방법
    33.
    发明公开
    음성신호에서 통계적 모델을 이용한 잡음 제거 장치 및 방법 无效
    通过在语音信号中使用统计模型来滤波噪声的装置及其方法

    公开(公告)号:KR1020110024969A

    公开(公告)日:2011-03-09

    申请号:KR1020090083167

    申请日:2009-09-03

    CPC classification number: G10L21/0208 G10L19/038 G10L19/26

    Abstract: PURPOSE: An apparatus for filtering a noise using a statistical model in a voice signal is provided to improve the wiener filter performance by restoring voice signal using a joint density GMM(Gaussian Mixture Model). CONSTITUTION: An apparatus for filtering a noise using a statistical model comprises a clean signal spectrum vector estimating unit(214), a post SNR estimating unit(216), a transfer function estimating unit(218), and a noise filtering unit(220). The clean signal spectrum vector estimating unit estimates spectrum vector of a clean signal using a PSD(Power Spectrum Density), a PSD estimation information of the estimated input signal, and a preset statistical model. The noise filtering unit performs noise filtering using the transfer function and fast fourier transformed frequency axis complex signal.

    Abstract translation: 目的:提供一种使用语音信号中的统计模型对噪声进行滤波的装置,以通过使用关节密度GMM(高斯混合模型)恢复语音信号来提高维纳滤波器性能。 构成:使用统计模型对噪声进行滤波的装置包括干净信号频谱矢量估计单元(214),后SNR估计单元(216),传递函数估计单元(218)和噪声滤波单元(220) 。 清洁信号频谱矢量估计单元使用PSD(功率谱密度)估计干净信号的频谱矢量,估计输入信号的PSD估计信息和预设的统计模型。 噪声滤波单元使用传递函数和快速傅立叶变换频率轴复信号进行噪声滤波。

    IPTV 방송 서비스 방법 및 서버와 IPTV 셋탑장치
    34.
    发明公开
    IPTV 방송 서비스 방법 및 서버와 IPTV 셋탑장치 无效
    方法和服务器服务广播IPTV和IPTV设置顶级设备

    公开(公告)号:KR1020100073172A

    公开(公告)日:2010-07-01

    申请号:KR1020080131766

    申请日:2008-12-22

    CPC classification number: H04N21/232 G06F17/30002 G10L15/02 H04N21/43

    Abstract: PURPOSE: A method and a server servicing IPTV broadcast and an IPTV set top apparatus are provided to search the large amount of IPTV broadcasting data by searching broadcasting data in real time through voice speech. CONSTITUTION: A voice recognition list transceiver(103) transmits and receives voice recognition list request information of an update scheduling unit(101). A voice recognition unit(105) recognizes a voice signal in an IPTV set-top box(100), and a voice search request transmitter(106) transmits a recognized string to a voice search request receiver. A voice search result receiver(107) provides the broadcasting data of a broadcasting data transmitter to a display unit(108).

    Abstract translation: 目的:提供一种服务于IPTV广播的方法和服务器,以及IPTV机顶装置,通过语音语音实时搜索广播数据来搜索大量的IPTV广播数据。 构成:语音识别列表收发器(103)发送和接收更新调度单元(101)的语音识别列表请求信息。 语音识别单元(105)识别IPTV机顶盒(100)中的语音信号,语音搜索请求发送器(106)将识别的串发送到语音搜索请求接收器。 语音搜索结果接收器(107)将广播数据发送器的广播数据提供给显示单元(108)。

    음원 분리 방법 및 그 장치
    35.
    发明公开
    음원 분리 방법 및 그 장치 有权
    分离信号信号的方法及其设备

    公开(公告)号:KR1020100073167A

    公开(公告)日:2010-07-01

    申请号:KR1020080131761

    申请日:2008-12-22

    CPC classification number: H04R3/005 H04R27/00 H04R2430/03

    Abstract: PURPOSE: A method for separating a source signals and an apparatus thereof are provided to improve the recording, transmission and recognition performances by separating only desirable sound source signal in plural sound source environments. CONSTITUTION: Fourier transformer(10) transforms a mixed input signal(S1) into each channel frequency domain through Fourier transformation. A frequency bandwidth divider(20) constitutes a frequency cluster from the each frequency domain. A frequency domain signal divider(30) applies a blind source separation for each cluster frequency domain. A reverse Fourier transformer(40) integrates the spectrums of divided signals through reverse Fourier transformation.

    Abstract translation: 目的:提供一种用于分离源信号的方法及其装置,以通过在多个声源环境中分离所需的声源信号来提高记录,传输和识别性能。 构成:傅里叶变换器(10)通过傅里叶变换将混合输入信号(S1)转换成每个通道频域。 频率带宽分配器(20)构成来自每个频域的频率簇。 频域信号分频器(30)为每个群集频域应用盲源分离。 反傅里叶变换器(40)通过反傅里叶变换对分频信号的频谱进行积分。

    음질 향상 장치와 음성 인식 시스템 및 방법
    36.
    发明公开
    음질 향상 장치와 음성 인식 시스템 및 방법 有权
    语音改进装置和语音识别系统及方法

    公开(公告)号:KR1020100072842A

    公开(公告)日:2010-07-01

    申请号:KR1020080131369

    申请日:2008-12-22

    CPC classification number: G10L21/0208 G10L15/20 G10L25/48

    Abstract: PURPOSE: A speech improving apparatus and a speech recognition system and method are provided to improve the voice recognition performance of a voice recognition system in a movable body of small resources by performing signal decoding through a sound model database. CONSTITUTION: A speed level divider(100) measures a moving speed level of a movable body through an inputted noise signal inputted in an initial stage of voice recognition. When the speed level of the movable body is lower than a predetermined value, a first sound quality improvement unit(112) improves the sound quality of a voice signal inputted by a Wiener filter. If the speed level of the movable body exceeds a predetermined value, a second sound quality improvement unit(114) improves the sound quality of a voice signal inputted by a GMM(Gaussian Mixture Model).

    Abstract translation: 目的:提供语音改善装置和语音识别系统和方法,通过声音模型数据库执行信号解码来提高小资源移动体中语音识别系统的语音识别性能。 构成:速度分级器(100)通过在语音识别的初始阶段输入的输入噪声信号测量可移动体的移动速度水平。 当可移动体的速度水平低于预定值时,第一音质改善单元(112)提高了由维纳滤波器输入的语音信号的声音质量。 如果可移动体的速度水平超过预定值,则第二音质改善单元(114)提高了由GMM(高斯混合模型)输入的语音信号的声音质量。

    캡스트럼 평균 차감 방법 및 그 장치
    37.
    发明公开
    캡스트럼 평균 차감 방법 및 그 장치 失效
    CEPSTRUM MEAN SUBTRACTION METHOD AND IET APPARATUS

    公开(公告)号:KR1020100069117A

    公开(公告)日:2010-06-24

    申请号:KR1020080127707

    申请日:2008-12-16

    Abstract: PURPOSE: A CMS(Cepstrum Mean Subtraction) method and a device thereof are provided to accurately normalize a channel property by estimating an average CMS value of the real voice section based on the CMS average value of a mute section. CONSTITUTION: A property extractor(200) extracts the properties of a mute section before a start point, a sound section, and a mute section after a finish point. A firing unit CMS value calculator(600) calculates an actual firing unit cepstrum average about the entire sound section. A cepstrum average estimator(300) estimates the cepstrum average of the entire section based on the properties of the mute section. A property vector CMS applier(400) performs channel-normalization of the estimated average. A decoder decodes the channel-normalized MFCC property vector.

    Abstract translation: 目的:提供CMS(倒谱平均减法)方法及其装置,以通过基于静音部分的CMS平均值估计真实语音部分的平均CMS值来准确地规范信道特性。 规定:属性提取器(200)在完成点之后提取起始点,声音部分和静音部分之前的静音部分的属性。 点火单元CMS值计算器(600)计算关于整个声音部分的实际发射单位倒谱平均值。 倒谱平均估计器(300)基于静音部分的属性来估计整个部分的倒谱平均值。 属性向量CMS应用程序(400)执行估计平均值的信道归一化。 解码器解码信道归一化的MFCC属性向量。

    차량용 네비게이션 단말기의 음성인식 방법
    38.
    发明公开
    차량용 네비게이션 단말기의 음성인식 방법 失效
    提供车辆导航系统中语音识别的方法

    公开(公告)号:KR1020100066917A

    公开(公告)日:2010-06-18

    申请号:KR1020080125434

    申请日:2008-12-10

    Abstract: PURPOSE: A voice recognition method of a vehicle navigation terminal is provided to generate voice emitting isoform through a simple pattern construction using a resolute/tagged result by presenting a meaning classification system for POI name domain. CONSTITUTION: A voice recognition method of a vehicle navigation terminal is as follows. The points of interest(POI) list and POI learning data are recognized from the voice information of a voice emitting isoform input to the vehicle navigation terminal (S200). A resource is built on the POI list and the POI learning data recognized(S202). The resolution and tagging on the built resource are performed with the POI list(S204). The result resolved and tagged is created as POI database(S206). Simplex/analyzed database is built based on the POI list and the POI learning data. N-gram vocabulary is extracted from the POI learning data.

    Abstract translation: 目的:提供一种车载导航终端的语音识别方法,通过呈现POI名称域的意义分类系统,通过简单的模式构造,通过坚决/标记的结果生成语音发射同种型。 构成:车辆导航终端的声音识别方法如下。 通过输入到车辆导航终端的发音同步体的语音信息来识别兴趣点(POI)列表和POI学习数据(S200)。 资源建立在POI列表和POI学习数据识别(S202)上。 使用POI列表执行内置资源的分辨率和标记(S204)。 解决和标记的结果被创建为POI数据库(S206)。 基于POI列表和POI学习数据构建S​​implex /分析数据库。 从POI学习数据中提取N-gram词汇表。

    혼동 행렬 기반 발화 검증 방법 및 장치
    39.
    发明授权
    혼동 행렬 기반 발화 검증 방법 및 장치 失效
    혼동행렬기반발화검증방법및장치

    公开(公告)号:KR100930587B1

    公开(公告)日:2009-12-09

    申请号:KR1020070122185

    申请日:2007-11-28

    Abstract: A confusion matrix based utterance verification method and an apparatus thereof are provided to select a phoneme with high discrimination by using a probability value of a confusion matrix as a weight for a likelihood value of a mono phone model. By performing viterbi decoding by using a context dependent phoneme mode, an inputted voice is recognized(307). A likelihood value of each phoneme, included in a pre-trained context independence phoneme model, and each phoneme, included in the voice-recognized character string as a voice recognition result, is calculated(309). Reliability for the voice-recognized character string is measured based on the calculated likelihood value of each phoneme and the pre-calculated probability value of the confusion matrix(311). It is determined whether to grant or reject the voice-recognized character string based on the measured reliability(313,315,317).

    Abstract translation: 提供基于混淆矩阵的发声验证方法及其装置,以通过使用混淆矩阵的概率值作为单声道手机型号似然值的权重来选择具有高判别度的音素。 通过使用上下文相关音素模式进行维特比解码,识别输入的语音(307)。 计算(309)包括在预先训练的上下文独立音素模型中的每个音素的似然值以及包括在作为语音识别结果的语音识别字符串中的每个音素。 基于计算出的每个音素的似然值和混淆矩阵的预先计算的概率值来测量语音识别字符串的可靠性(311)。 基于测量的可靠性来确定是否授予或拒绝语音识别字符串(313,315,317)。

    잡음 제거 장치 및 방법
    40.
    发明公开
    잡음 제거 장치 및 방법 无效
    用于减少噪声的装置和方法

    公开(公告)号:KR1020090111739A

    公开(公告)日:2009-10-27

    申请号:KR1020080075653

    申请日:2008-08-01

    Abstract: PURPOSE: A noise cancelling apparatus is provided to estimate a clean voice more accurately in an environment in which a dynamic noise and various noises are mixed. CONSTITUTION: A noise cancelling apparatus comprises a noise estimation module(200) which calculates the estimation value of a noise signal in the current frame of a voice signal, a Wiener filter module(202) which receives the voice signal and calculates an intermediate result by applying the intermediate Wiener filter, a database(206) in which Gaussian mixed-model data is stored, and an MMSE estimation module(204) which calculates the estimation value of a clean voice by using the Gaussian mixed-model data and intermediate result.

    Abstract translation: 目的:提供一种噪声消除装置,用于在动态噪声和各种噪声混合的环境中更精确地估计干净的声音。 噪声消除装置包括噪声估计模块(200),噪声估计模块(200),其计算语音信号的当前帧中的噪声信号的估计值;维纳滤波器模块(202),其接收语音信号并通过以下步骤计算中间结果 应用中间维纳滤波器,存储高斯混合模型数据的数据库(206)和通过使用高斯混合模型数据和中间结果来计算干净声音的估计值的MMSE估计模块(204)。

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