CN108172235B - 基于维纳后置滤波的ls波束形成混响抑制方法 - Google Patents
基于维纳后置滤波的ls波束形成混响抑制方法 Download PDFInfo
- Publication number
- CN108172235B CN108172235B CN201711431478.XA CN201711431478A CN108172235B CN 108172235 B CN108172235 B CN 108172235B CN 201711431478 A CN201711431478 A CN 201711431478A CN 108172235 B CN108172235 B CN 108172235B
- Authority
- CN
- China
- Prior art keywords
- signal
- frequency
- microphone
- response
- reverberation
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
- 238000000034 method Methods 0.000 title claims abstract description 24
- 230000001629 suppression Effects 0.000 title claims abstract description 20
- 238000001914 filtration Methods 0.000 title claims abstract description 19
- 230000004044 response Effects 0.000 claims description 27
- 239000013598 vector Substances 0.000 claims description 22
- 238000005316 response function Methods 0.000 claims description 9
- 238000001228 spectrum Methods 0.000 claims description 6
- 230000000694 effects Effects 0.000 claims description 5
- 239000011159 matrix material Substances 0.000 claims description 5
- 238000005311 autocorrelation function Methods 0.000 claims description 4
- 230000008569 process Effects 0.000 claims description 3
- 238000009795 derivation Methods 0.000 claims description 2
- 238000013461 design Methods 0.000 claims description 2
- 238000012545 processing Methods 0.000 abstract description 5
- 230000006870 function Effects 0.000 description 7
- 238000010586 diagram Methods 0.000 description 3
- 238000004458 analytical method Methods 0.000 description 2
- 238000003491 array Methods 0.000 description 2
- 238000004891 communication Methods 0.000 description 2
- 230000003111 delayed effect Effects 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 239000000654 additive Substances 0.000 description 1
- 230000000996 additive effect Effects 0.000 description 1
- 230000009286 beneficial effect Effects 0.000 description 1
- 230000007547 defect Effects 0.000 description 1
- 230000006872 improvement Effects 0.000 description 1
- 238000013178 mathematical model Methods 0.000 description 1
- 230000000644 propagated effect Effects 0.000 description 1
- 230000009467 reduction Effects 0.000 description 1
- 238000011160 research Methods 0.000 description 1
- 230000005236 sound signal Effects 0.000 description 1
- 230000003595 spectral effect Effects 0.000 description 1
Images
Classifications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
Landscapes
- Engineering & Computer Science (AREA)
- Computational Linguistics (AREA)
- Quality & Reliability (AREA)
- Signal Processing (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Circuit For Audible Band Transducer (AREA)
Abstract
本发明提供了一种具有维纳后置滤波的最小二乘波束形成混响抑制方法。该发明算法将混响后的语音信号分为直达部分和混响部分得到维纳后置滤波器增益估计;针对语音信号在低频部分噪声相干性较强,利用最小二乘波束形成算法进行分频处理,最后求解最优权值。优点:将混响后的信号分为直达部分和混响部分得到改进维纳后置滤波器增益估计,并针对语音信号在低频部分噪声相干性较强的特点,将混响后的语音信号分为高频和低频分量,然后用最小二乘波束形成算法分别求解高、低频分量的最优权值,提高混响抑制精度和语音质量。
Description
技术领域
本发明涉及一种基于维纳后置滤波的最小二乘(LS)波束形成混响抑制方法,属于麦克风阵列波束形成技术领域。
背景技术
麦克风阵列是语音获取的有效工具,它被广泛地应用于语音识别、视频会议助听设备等。波束形成是一种重要的麦克风阵列处理技术。近年随着人们对语音通信研究的深入,麦克风阵列得到更加广泛的应用。
封闭空间环境中的语音信号经常被混响所扭曲。在具有多个分布式麦克风的语音通信应用中,通常期望量化每个传感器处感知信号的混响量,以便选择具有最高质量或最小混响的频道。假设不同信道上的噪声之间不相关的前提下,R.Zelinski提出具有维纳后置滤波的波束形成器,利用空间信息解决了维纳滤波器的估计问题。但这种非相干噪声场实际上很少遇到,特别是低频噪声场。Berkun和Claude Marro提出基于麦克阵列与维纳后置滤波器结合的降噪和去混响算法。McCowan运用扩散噪声场的数学模型讨论了解决扩散噪声场中不同通道噪声相关的问题,该算法要求预先得到噪声相干函数,适用范围受到限制。K.U.Simmer提出的多通道维纳滤波器(MCWF),其可以分解为最小方差无失真响应波束形成器和单通道后置滤波器,求最优解表达式,对混响中语音质量改善明显。AlejandroLuebs在白噪声和漫反射噪声的基础上增加点干扰处理,通过提供全局最优的最小二乘解决方案,更有效地利用麦克风阵列收集的信息,提高语音质量。
发明内容
本发明所要解决的技术问题是克服现有技术的缺陷,提供一种基于维纳后置滤波的LS波束形成混响抑制方法,其特征在于,麦克风的接收信号x(t)经过维纳后置滤波的最小二乘波束形成混响抑制方法处理得到的输出信号:y(t)=WHx(t),其中,W表示麦克风阵列响应的权矢量,()H表示共轭转置,t表示时间序列,表示t时刻第m麦克风的接收信号,M为麦克风阵元数目,L为房间冲击响应长度,G为房间冲击响应,sm(t)为t时刻第m麦克风采集的纯净语音信号。
进一步的,所述接收信号x(t)=[x1(t),x2(t),…,xM(t)]。
进一步的,所述麦克风阵列响应权矢量W的获取,步骤如下:
步骤a:在波束形成器设计时应用最小二乘波束形成算法,将目标函数定义为式中n,k分别表示角度和频率的离散点数;Nφ、Nf分别为角度和频率范围,Fnk为正实值的加权函数,Ynk为实际波束形成器响应函数,ank是空时二维导向矢量,Dnk为期望波束响应,h为波束形成器权矢量;
步骤c:基于房间脉冲响应h(k)是一个随机过程,表示为式中b(k)为零均值的高斯白噪声,Δ是与混响时间T60相联系的衰减因子,从房间脉冲响应h(k)的角度看,可以把h(k)近似分成直达部分语音信号的脉冲响应函数hd(k)和形成混响信号的响应函数hr(k),β是本位设定的临界时间,房间冲击响应在k<β时的混响效应不明显,与干净语音的卷积可以看作直达声,分别表示为假设sd(k)与sr(k)分别表示纯净语音信号s(t)与hd(k)和hr(k)的卷积,则sd(k)为待处理语音信号的直达信号部分,sr(k)为待处理语音信号的混响部分,得到改进维纳滤波器的估计增益式中为直达信号的自相关函数,为麦克风接收信号的自相关函数,E[]为取均值,R[]为取实部,M为麦克风阵元数目,下标i,j是麦克风通道标号。
步骤d:根据步骤a,b,c得到基于维纳后置滤波的LS改进波束形成混响抑制方法,麦克风阵列响应权矢量α为加权矩阵系数,hL,hH分别表示信号在低频和高频波束形成器权矢量,高频和低频分量的频率分界点取为1kHz。
进一步的,所述步骤c中的本位设定的临界时间β=50ms。
进一步的,所述步骤d,将步骤a中波束形成器权矢量h,以1kHz为高频和低频分量的频率分界点划分为hL,hH,α为加权矩阵系数,将高低频权矢量分别相加,即αhL+(1-α)hH;再与步骤c中改进维纳滤波器的估计增益相乘,得到麦克风阵列响应权矢量
本发明所达到的有益效果:为了提高封闭空间环境中麦克风阵列接收的语音信号质量,提出一种具有维纳后置滤波的最小二乘波束形成混响抑制方法。算法将混响后的信号分为直达部分和混响部分得到改进维纳后置滤波器增益估计,并针对语音信号在低频部分噪声相干性较强的特点,将混响后的语音信号分为高频和低频分量,然后用最小二乘波束形成算法分别求解高、低频分量的最优权值,提高混响抑制精度和语音质量。
附图说明
图1是本发明的基于维纳后置滤波的LS波束形成混响抑制方法原理图;
图2是纯净语音信号语谱图;
图3是混响后信号语谱图;
图4是本发明的算法去混响语谱图。
具体实施方式
下面结合附图对本发明作进一步描述。以下实施例仅用于更加清楚地说明本发明的技术方案,而不能以此来限制本发明的保护范围。
图1是基于维纳后置滤波的LS波束形成混响抑制方法原理图,在图1中由M个相同的全向性麦克风组成均匀线阵,有N个语音信号(M>N)。
步骤1、假设麦克风采集的信号都是延迟和衰减之后的原始语音信号加上一定的加性噪声。则第m个麦克风接收的信号xm(k)=αmsn(k)+vm(k),其中,αm,m=1,…M表示传播效应引起的衰减因子;sn(k),n=1,…N是第n个语音到第m个麦克风的语音信号;vm(k)表示第m个麦克风接收的噪声信号,k是离散时间。假设在封闭的室内环境下,第m个麦克风接收的信号可以表示为式中Gnm,l是第n个语音到第m个麦克风,长度为l的房间冲激响应,且m=1,…M;n=1,…N;l=1,…L。由于语音信号的动态非平稳特性,对采用傅里叶变换(FFT)得,sn(ω,k)表示sn(k)第k帧信号短时谱。
步骤2、随着特定房间对不同频率的衰减和反射程度而改变的,即不同频率的声信号产生的混响有一定的差异,并且在实际声场中低频部分噪声相干性较强,因此采用分频处理的思想,将傅里叶变换后的信号分为高频和低频分量,频率分界点取为1kHz。将分频后的信号,用LS波束形成算法分别进行处理后再求和,将得到的信号Y(ω)进行维纳后置滤波。
在最小二乘波束形成器的设计方法中,将目标函数定义为
将式(1)目标函数展开,并缩写为
则式(1)可写为
J(h)=hTRh-2qTh+dLS (4)
h=R-1q (5)
步骤3、在封闭环境内,麦克风阵列采集到的信号不仅包含直达路径传播的信号,而且包含了由于房间反射而产生的延迟衰减信号,这种多径传播效应在接收信号中导致谱失真,称为混响,混响后的语音信号语谱图为图3。基于房间脉冲响应h(k)是一个随机过程,表示为
式中b(k)为零均值的高斯白噪声,Δ是与混响时间T60相联系的衰减因子,从房间脉冲响应h(k)的角度看,可以把h(k)近似分成直达部分语音信号的脉冲响应函数hd(k)和形成混响信号的响应函数hr(k),β=50ms是本位设定的临界时间,房间冲击响应在k<50ms时的混响效应不明显,与干净语音的卷积可以看作直达声。分别表示为
假设sd(k)与sr(k)分别表示纯净语音信号s(t)与hd(k)和hr(k)的卷积,则sd(k)为待处理语音信号的直达信号部分,sr(k)为待处理语音信号的混响部分。由以上分析得到改进维纳滤波器的估计增益。
以上所述仅是本发明的优选实施方式,应当指出,对于本技术领域的普通技术人员来说,在不脱离本发明技术原理的前提下,还可以做出若干改进和变形,这些改进和变形也应视为本发明的保护范围。
Claims (5)
1.一种基于维纳后置滤波的LS波束形成混响抑制方法,其特征在于,麦克风的接收信号x(t)经过维纳后置滤波的最小二乘波束形成混响抑制方法处理得到的输出信号:y(t)=WHx(t),其中,W表示麦克风阵列响应的权矢量,()H表示共轭转置,t表示时间序列,表示t时刻第m麦克风的接收信号,M为麦克风阵元数目,L为房间冲击响应长度,G为房间冲击响应,sm(t)为t时刻第m麦克风采集的纯净语音信号;
所述麦克风阵列响应权矢量W的获取,步骤如下:
步骤a:在波束形成器设计时应用最小二乘波束形成算法,将目标函数定义为式中n,k分别表示角度和频率的离散点数;Nφ、Nf分别为角度和频率范围,Fnk为正实值的加权函数,Ynk为实际波束形成器响应函数,ank是空时二维导向矢量,Dnk为期望波束响应,h为波束形成器权矢量;
步骤c:基于房间脉冲响应h(k)是一个随机过程,表示为式中b(k)为零均值的高斯白噪声,Δ是与混响时间T60相联系的衰减因子,从房间脉冲响应h(k)的角度看,可以把h(k)近似分成直达部分语音信号的脉冲响应函数hd(k)和形成混响信号的响应函数hr(k),β是本位设定的临界时间,房间冲击响应在k<β时的混响效应不明显,与干净语音的卷积可以看作直达声,分别表示为假设sd(k)与sr(k)分别表示纯净语音信号s(t)与hd(k)和hr(k)的卷积,则sd(k)为待处理语音信号的直达信号部分,sr(k)为待处理语音信号的混响部分,得到改进维纳滤波器的估计增益式中为直达信号的自相关函数,为麦克风接收信号的自相关函数,E[]为取均值,R[]为取实部,M为麦克风阵元数目,下标i,j是麦克风通道标号;
2.根据权利要求1所述的基于维纳后置滤波的LS波束形成混响抑制方法,其特征在于,所述接收信号x(t)=[x1(t),x2(t),…,xM(t)]。
3.根据权利要求1所述的基于维纳后置滤波的LS波束形成混响抑制方法,其特征在于,所述步骤c中的本位设定的临界时间β=50ms。
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN201711431478.XA CN108172235B (zh) | 2017-12-26 | 2017-12-26 | 基于维纳后置滤波的ls波束形成混响抑制方法 |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN201711431478.XA CN108172235B (zh) | 2017-12-26 | 2017-12-26 | 基于维纳后置滤波的ls波束形成混响抑制方法 |
Publications (2)
Publication Number | Publication Date |
---|---|
CN108172235A CN108172235A (zh) | 2018-06-15 |
CN108172235B true CN108172235B (zh) | 2021-05-14 |
Family
ID=62521178
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
CN201711431478.XA Active CN108172235B (zh) | 2017-12-26 | 2017-12-26 | 基于维纳后置滤波的ls波束形成混响抑制方法 |
Country Status (1)
Country | Link |
---|---|
CN (1) | CN108172235B (zh) |
Families Citing this family (19)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US9565493B2 (en) | 2015-04-30 | 2017-02-07 | Shure Acquisition Holdings, Inc. | Array microphone system and method of assembling the same |
US9554207B2 (en) | 2015-04-30 | 2017-01-24 | Shure Acquisition Holdings, Inc. | Offset cartridge microphones |
US10367948B2 (en) | 2017-01-13 | 2019-07-30 | Shure Acquisition Holdings, Inc. | Post-mixing acoustic echo cancellation systems and methods |
US11523212B2 (en) | 2018-06-01 | 2022-12-06 | Shure Acquisition Holdings, Inc. | Pattern-forming microphone array |
US11297423B2 (en) | 2018-06-15 | 2022-04-05 | Shure Acquisition Holdings, Inc. | Endfire linear array microphone |
WO2020061353A1 (en) | 2018-09-20 | 2020-03-26 | Shure Acquisition Holdings, Inc. | Adjustable lobe shape for array microphones |
US11558693B2 (en) | 2019-03-21 | 2023-01-17 | Shure Acquisition Holdings, Inc. | Auto focus, auto focus within regions, and auto placement of beamformed microphone lobes with inhibition and voice activity detection functionality |
WO2020191380A1 (en) | 2019-03-21 | 2020-09-24 | Shure Acquisition Holdings,Inc. | Auto focus, auto focus within regions, and auto placement of beamformed microphone lobes with inhibition functionality |
US11303981B2 (en) | 2019-03-21 | 2022-04-12 | Shure Acquisition Holdings, Inc. | Housings and associated design features for ceiling array microphones |
WO2020237206A1 (en) | 2019-05-23 | 2020-11-26 | Shure Acquisition Holdings, Inc. | Steerable speaker array, system, and method for the same |
WO2020243471A1 (en) | 2019-05-31 | 2020-12-03 | Shure Acquisition Holdings, Inc. | Low latency automixer integrated with voice and noise activity detection |
US11297426B2 (en) | 2019-08-23 | 2022-04-05 | Shure Acquisition Holdings, Inc. | One-dimensional array microphone with improved directivity |
WO2021087377A1 (en) | 2019-11-01 | 2021-05-06 | Shure Acquisition Holdings, Inc. | Proximity microphone |
CN111462770A (zh) * | 2020-01-09 | 2020-07-28 | 华中科技大学 | 一种基于lstm的后期混响抑制方法及系统 |
US11552611B2 (en) | 2020-02-07 | 2023-01-10 | Shure Acquisition Holdings, Inc. | System and method for automatic adjustment of reference gain |
WO2021243368A2 (en) | 2020-05-29 | 2021-12-02 | Shure Acquisition Holdings, Inc. | Transducer steering and configuration systems and methods using a local positioning system |
EP4285605A1 (en) | 2021-01-28 | 2023-12-06 | Shure Acquisition Holdings, Inc. | Hybrid audio beamforming system |
CN113724723B (zh) * | 2021-09-02 | 2024-06-11 | 西安讯飞超脑信息科技有限公司 | 混响与噪声抑制方法、装置、电子设备及存储介质 |
CN114267371A (zh) * | 2021-12-30 | 2022-04-01 | 思必驰科技股份有限公司 | 去混响方法、电子设备和存储介质 |
Family Cites Families (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US20160240210A1 (en) * | 2012-07-22 | 2016-08-18 | Xia Lou | Speech Enhancement to Improve Speech Intelligibility and Automatic Speech Recognition |
CN104835503A (zh) * | 2015-05-06 | 2015-08-12 | 南京信息工程大学 | 一种改进gsc自适应语音增强方法 |
US9721582B1 (en) * | 2016-02-03 | 2017-08-01 | Google Inc. | Globally optimized least-squares post-filtering for speech enhancement |
CN106782590B (zh) * | 2016-12-14 | 2020-10-09 | 南京信息工程大学 | 基于混响环境下麦克风阵列波束形成方法 |
-
2017
- 2017-12-26 CN CN201711431478.XA patent/CN108172235B/zh active Active
Also Published As
Publication number | Publication date |
---|---|
CN108172235A (zh) | 2018-06-15 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
CN108172235B (zh) | 基于维纳后置滤波的ls波束形成混响抑制方法 | |
CN105590631B (zh) | 信号处理的方法及装置 | |
CN106782590B (zh) | 基于混响环境下麦克风阵列波束形成方法 | |
CN106710601B (zh) | 一种语音信号降噪拾音处理方法和装置及冰箱 | |
EP2936830B1 (en) | Filter and method for informed spatial filtering using multiple instantaneous direction-of-arrivial estimates | |
CN105869651B (zh) | 基于噪声混合相干性的双通道波束形成语音增强方法 | |
WO2015196729A1 (zh) | 一种麦克风阵列语音增强方法及装置 | |
CN106251877A (zh) | 语音声源方向估计方法及装置 | |
KR101834913B1 (ko) | 복수의 입력 오디오 신호를 잔향제거하기 위한 신호 처리 장치, 방법 및 컴퓨터가 판독 가능한 저장매체 | |
US20140025374A1 (en) | Speech enhancement to improve speech intelligibility and automatic speech recognition | |
Schwartz et al. | Joint estimation of late reverberant and speech power spectral densities in noisy environments using Frobenius norm | |
Ito et al. | Designing the Wiener post-filter for diffuse noise suppression using imaginary parts of inter-channel cross-spectra | |
EP3245795A2 (en) | Reverberation suppression using multiple beamformers | |
Schwartz et al. | Joint maximum likelihood estimation of late reverberant and speech power spectral density in noisy environments | |
CN106031196A (zh) | 信号处理装置、方法以及程序 | |
CN110111802B (zh) | 基于卡尔曼滤波的自适应去混响方法 | |
CN114255777A (zh) | 实时语音去混响的混合方法及系统 | |
Kumatani et al. | Microphone array post-filter based on spatially-correlated noise measurements for distant speech recognition | |
Modhave et al. | Design of multichannel wiener filter for speech enhancement in hearing aids and noise reduction technique | |
JP2017181761A (ja) | 信号処理装置及びプログラム、並びに、ゲイン処理装置及びプログラム | |
CN111210836A (zh) | 一种麦克风阵列波束形成动态调整方法 | |
Aichner et al. | Least-squares error beamforming using minimum statistics and multichannel frequency-domain adaptive filtering | |
Li et al. | Subband gradient flow acoustic source separation for moderate reverberation environment | |
Hofbauer et al. | Limitations for FIR multi-microphone speech dereverberation in the low-delay case | |
Zhao et al. | A multichannel widely linearwiener filter for binaural noise reduction in the short-time-fourier-transform domain |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
PB01 | Publication | ||
SE01 | Entry into force of request for substantive examination | ||
SE01 | Entry into force of request for substantive examination | ||
GR01 | Patent grant | ||
GR01 | Patent grant |