Model-based Noisy Speech Recognition with Environment Parameters Estimated by Noise Adaptiv
外文翻译---说话人识别
附录A 英文文献Speaker RecognitionBy Judith A. Markowitz, J. Markowitz ConsultantsSpeaker recognition uses features of a person‟s voice to identify or verify that person. It is a well-established biometric with commercial systems that are more than 10 years old and deployed non-commercial systems that are more than 20 years old. This paper describes how speaker recognition systems work and how they are used in applications.1. IntroductionSpeaker recognition (also called voice ID and voice biometrics) is the only human-biometric technology in commercial use today that extracts information from sound patterns. It is also one of the most well-established biometrics, with deployed commercial applications that are more than 10 years old and non-commercial systems that are more than 20 years old.2. How do Speaker-Recognition Systems WorkSpeaker-recognition systems use features of a person‟s voice and speaking style to:●attach an identity to the voice of an unknown speaker●verify that a person is who she/ he claims to be●separate one person‟s voice from other voices in a multi-speakerenvironmentThe first operation is called speak identification or speaker recognition; the second has many names, including speaker verification, speaker authentication, voice verification, and voice recognition; the third is speaker separation or, in some situations, speaker classification. This papers focuses on speaker verification, the most highly commercialized of these technologies.2.1 Overview of the ProcessSpeaker verification is a biometric technology used for determining whether the person is who she or he claims to be. It should not be confused with speech recognition, a non-biometric technology used for identifying what a person is saying. Speech recognition products are not designed to determine who is speaking.Speaker verification begins with a claim of identity (see Figure A1). Usually, the claim entails manual entry of a personal identification number (PIN), but a growing number of products allow spoken entry of the PIN and use speech recognition to identify the numeric code. Some applications replace manual or spoken PIN entry with bank cards, smartcards, or the number of the telephone being used. PINS are also eliminated when a speaker-verification system contacts the user, an approach typical of systems used to monitor home-incarcerated criminals.Figure A1.Once the identity claim has been made, the system retrieves the stored voice sample (called a voiceprint) for the claimed identity and requests spoken input from the person making the claim. Usually, the requested input is a password. The newly input speech is compared with the stored voiceprint and the results of that comparison are measured against an acceptance/rejection threshold. Finally, the system accepts the speaker as the authorized user, rejects the speaker as an impostor, or takes another action determined by the application. Some systems report a confidence level or other score indicating how confident it about its decision.If the verification is successful the system may update the acoustic information in the stored voiceprint. This process is called adaptation. Adaptation is an unobtrusive solution for keeping voiceprints current and is used by many commercial speaker verification systems.2.2 The Speech SampleAs with all biometrics, before verification (or identification) can be performed the person must provide a sample of speech (called enrolment). The sample is used to create the stored voiceprint.Systems differ in the type and amount of speech needed for enrolment and verification. The basic divisions among these systems are●text dependent●text independent●text prompted2.2.1 Text DependentMost commercial systems are text dependent.Text-dependent systems expect the speaker to say a pre-determined phrase, password, or ID. By controlling the words that are spoken the system can look for a close match with the stored voiceprint. Typically, each person selects a private password, although some administrators prefer to assign passwords. Passwords offer extra security, requiring an impostor to know the correct PIN and password and to have a matching voice. Some systems further enhance security by not storing a human-readable representation of the password.A global phrase may also be used. In its 1996 pilot of speaker verification Chase Manhattan Bank used …Verification by Chemical Bank‟. Global phrases avoid the problem of forgotten passwords, but lack the added protection offered by private passwords.2.2.2 Text IndependentText-independent systems ask the person to talk. What the person says is different every time. It is extremely difficult to accurately compare utterances that are totally different from each other - particularly in noisy environments or over poor telephone connections. Consequently, commercial deployment of text-independentverification has been limited.2.2.3 Text PromptedText-prompted systems (also called challenge response) ask speakers to repeat one or more randomly selected numbers or words (e.g. “43516”, “27,46”, or “Friday, c omputer”). Text prompting adds time to enrolment and verification, but it enhances security against tape recordings. Since the items to be repeated cannot be predicted, it is extremely difficult to play a recording. Furthermore, there is no problem of forgetting a password, even though the PIN, if used, may still be forgotten.2.3 Anti-speaker ModellingMost systems compare the new speech sample with the stored voiceprint for the claimed identity. Other systems also compare the newly input speech with the voices of other people. Such techniques are called anti-speaker modelling. The underlying philosophy of anti-speaker modelling is that under any conditions a voice sample from a particular speaker will be more like other samples from that person than voice samples from other speakers. If, for example, the speaker is using a bad telephone connection and the match with the speaker‟s voiceprint is poor, it is likely that the scores for the cohorts (or world model) will be even worse.The most common anti-speaker techniques are●discriminate training●cohort modeling●world modelsDiscriminate training builds the comparisons into the voiceprint of the new speaker using the voices of the other speakers in the system. Cohort modelling selects a small set of speakers whose voices are similar to that of the person being enrolled. Cohorts are, for example, always the same sex as the speaker. When the speaker attempts verification, the incoming speech is compared with his/her stored voiceprint and with the voiceprints of each of the cohort speakers. World models (also called background models or composite models) contain a cross-section of voices. The same world model is used for all speakers.2.4 Physical and Behavioural BiometricsSpeaker recognition is often characterized as a behavioural biometric. This description is set in contrast with physical biometrics, such as fingerprinting and iris scanning. Unfortunately, its classification as a behavioural biometric promotes the misunderstanding that speaker recognition is entirely (or almost entirely) behavioural. If that were the case, good mimics would have no difficulty defeating speaker-recognition systems. Early studies determined this was not the case and identified mimic-resistant factors. Those factors reflect the size and shape of a speaker‟s speaking mechanism (called the vocal tract).The physical/behavioural classification also implies that performance of physical biometrics is not heavily influenced by behaviour. This misconception has led to the design of biometric systems that are unnecessarily vulnerable to careless and resistant users. This is unfortunate because it has delayed good human-factors design for those biometrics.3. How is Speaker Verification Used?Speaker verification is well-established as a means of providing biometric-based security for:●telephone networks●site access●data and data networksand monitoring of:●criminal offenders in community release programmes●outbound calls by incarcerated felons●time and attendance3.1 Telephone NetworksToll fraud (theft of long-distance telephone services) is a growing problem that costs telecommunications services providers, government, and private industry US$3-5 billion annually in the United States alone. The major types of toll fraud include the following:●Hacking CPE●Calling card fraud●Call forwarding●Prisoner toll fraud●Hacking 800 numbers●Call sell operations●900 number fraud●Switch/network hits●Social engineering●Subscriber fraud●Cloning wireless telephonesAmong the most damaging are theft of services from customer premises equipment (CPE), such as PBXs, and cloning of wireless telephones. Cloning involves stealing the ID of a telephone and programming other phones with it. Subscriber fraud, a growing problem in Europe, involves enrolling for services, usually under an alias, with no intention of paying for them.Speaker verification has two features that make it ideal for telephone and telephone network security: it uses voice input and it is not bound to proprietary hardware. Unlike most other biometrics that need specialized input devices, speaker verification operates with standard wireline and/or wireless telephones over existing telephone networks. Reliance on input devices created by other manufacturers for a purpose other than speaker verification also means that speaker verification cannot expect the consistency and quality offered by a proprietary input device. Speaker verification must overcome differences in input quality and the way in which speech frequencies are processed. This variability is produced by differences in network type (e.g. wireline v wireless), unpredictable noise levels on the line and in the background, transmission inconsistency, and differences in the microphone in telephone handset. Sensitivity to such variability is reduced through techniques such as speech enhancement and noise modelling, but products still need to be tested under expected conditions of use.Applications of speaker verification on wireline networks include secure calling cards, interactive voice response (IVR) systems, and integration with security forproprietary network systems. Such applications have been deployed by organizations as diverse as the University of Maryland, the Department of Foreign Affairs and International Trade Canada, and AMOCO. Wireless applications focus on preventing cloning but are being extended to subscriber fraud. The European Union is also actively applying speaker verification to telephony in various projects, including Caller Verification in Banking and Telecommunications, COST250, and Picasso.3.2 Site accessThe first deployment of speaker verification more than 20 years ago was for site access control. Since then, speaker verification has been used to control access to office buildings, factories, laboratories, bank vaults, homes, pharmacy departments in hospitals, and even access to the US and Canada. Since April 1997, the US Department of Immigration and Naturalization (INS) and other US and Canadian agencies have been using speaker verification to control after-hours border crossings at the Scobey, Montana port-of-entry. The INS is now testing a combination of speaker verification and face recognition in the commuter lane of other ports-of-entry.3.3 Data and Data NetworksGrowing threats of unauthorized penetration of computing networks, concerns about security of the Internet, and increases in off-site employees with data access needs have produced an upsurge in the application of speaker verification to data and network security.The financial services industry has been a leader in using speaker verification to protect proprietary data networks, electronic funds transfer between banks, access to customer accounts for telephone banking, and employee access to sensitive financial information. The Illinois Department of Revenue, for example, uses speaker verification to allow secure access to tax data by its off-site auditors.3.4 CorrectionsIn 1993, there were 4.8 million adults under correctional supervision in the United States and that number continues to increase. Community release programmes, such as parole and home detention, are the fastest growing segments of this industry. It is no longer possible for corrections officers to provide adequate monitoring ofthose people.In the US, corrections agencies have turned to electronic monitoring systems. Since the late 1980s speaker verification has been one of those electronic monitoring tools. Today, several products are used by corrections agencies, including an alcohol breathalyzer with speaker verification for people convicted of driving while intoxicated and a system that calls offenders on home detention at random times during the day.Speaker verification also controls telephone calls made by incarcerated felons. Inmates place a lot of calls. In 1994, US telecommunications services providers made $1.5 billion on outbound calls from inmates. Most inmates have restrictions on whom they can call. Speaker verification ensures that an inmate is not using another inmate‟s PIN to make a forbidden contact.3.5 Time and AttendanceTime and attendance applications are a small but growing segment of the speaker-verification market. SOC Credit Union in Michigan has used speaker verification for time and attendance monitoring of part-time employees for several years. Like many others, SOC Credit Union first deployed speaker verification for security and later extended it to time and attendance monitoring for part-time employees.4. StandardsThis paper concludes with a short discussion of application programming interface (API) standards. An API contains the function calls that enable programmers to use speaker-verification to create a product or application. Until April 1997, when the Speaker Verification API (SV API) standard was introduced, all available APIs for biometric products were proprietary. SV API remains the only API standard covering a specific biometric. It is now being incorporated into proposed generic biometric API standards. SV API was developed by a cross-section of speaker-recognition vendors, consultants, and end-user organizations to address a spectrum of needs and to support a broad range of product features. Because it supports both high level functions (e.g. calls to enrol) and low level functions (e.g. choices of audio input features) itfacilitates development of different types of applications by both novice and experienced developers.Why is it important to support API standards? Developers using a product with a proprietary API face difficult choices if the vendor of that product goes out of business, fails to support its product, or does not keep pace with technological advances. One of those choices is to rebuild the application from scratch using a different product. Given the same events, developers using a SV API-compliant product can select another compliant vendor and need perform far fewer modifications. Consequently, SV API makes development with speaker verification less risky and less costly. The advent of generic biometric API standards further facilitates integration of speaker verification with other biometrics. All of this helps speaker-verification vendors because it fosters growth in the marketplace. In the final analysis active support of API standards by developers and vendors benefits everyone.附录B 中文翻译说话人识别作者:Judith A. Markowitz, J. Markowitz Consultants 说话人识别是用一个人的语音特征来辨认或确认这个人。
语音识别中的声学模型适应研究
语音识别中的声学模型适应研究语音识别(Speech Recognition)是一种将语音信号转化为文本的技术。
在语音识别的过程中,声学模型(Acoustic Model)起着至关重要的作用。
声学模型是一种统计模型,它通过学习大量的语音数据,将声学特征与对应的文本进行关联,从而实现语音信号到文本的转换。
然而,在实际应用中,由于不同人群、不同环境等因素的影响,传统声学模型往往无法满足准确识别不同人群、不同环境下的语音信号。
因此,声学模型适应研究成为了当前研究领域中备受关注和重要性。
声学模型适应研究旨在通过对现有声学模型进行优化和改进,以适应各种复杂环境下的语音信号。
传统声学模型通常基于大量标准发音人数据进行训练,在实际使用中由于说话人差异、环境噪声等因素会导致识别准确率下降。
为了解决这一问题,研究者们提出了各种方法来改进和优化传统声学模型。
一种常用的方法是使用领域自适应技术(Domain Adaptation),该技术通过在特定领域收集和使用特定领域的语音数据,来优化声学模型。
领域自适应技术可以分为无监督和有监督两种。
无监督的领域自适应方法通过对齐不同语音数据的声学特征,来优化模型。
有监督的领域自适应方法则使用特定领域的标注数据来训练模型,以提高在该特定领域下的识别准确率。
此外,还有一种常用的方法是使用说话人自适应技术(Speaker Adaptation)。
说话人自适应技术通过对不同说话人语音数据进行建模和训练,以提高对不同说话人语音信号的识别准确率。
常见的说话人自适应方法包括基于高斯混合模型(GMM)和深度神经网络(DNN)等。
除了上述两种常用方法外,还有一些其他声学模型适应研究方法也值得关注。
例如,基于迁移学习(Transfer Learning)思想的声学模型优化方法可以通过在源任务上训练好一个初始声学模型,并将其迁移到目标任务上进行微调和优化。
此外,基于增量学习(Incremental Learning)的声学模型适应方法可以通过不断增加新的语音数据,来逐步优化声学模型。
基于LPC的混响时间估计算法
基于LPC的混响时间估计算法刘兴亮;姚剑敏;郭太良【摘要】准确计算混响时间需要知道房间的尺寸、墙壁的吸声系数等.经典的混响时间盲估计方法可以避免这些条件,但需要事先提供一个冲激信号.文章对经典算法进行了改进,提出了一种基于线性预测的混响时间盲估计算法.首先,将采集到的语音通过一个低阶的线性预测滤波器来获得线性预测残差信号;其次,计算残差信号的自相关,并选取合适的部分;最后,将选取的部分通过一个最大似然估计器,提取参数计算混响时间.文章还提出了一种改进的二分法来求解最大似然估计方程.实验证明,与经典算法相比,所提出算法估计的混响时间精度更高,且更具有实时性.%Accurate calculation of the reverberation time need to know the size of the room and the absorption characteristics of the walls.Classical method of blind reverberation time estimation can estimate the reverberation time without knowing the size of the room and the absorption characteristics of the walls,but have to provide an impulse signal.In this paper,we proposed a linear prediction based method.Firstly,the input speech is passed through a low order linear prediction coding filter to obtain the LP residual signal.Then,calculating the autocorrelation function of the LP residual signal and extracting the appropriate stly,calculating the reverberation time with the ML estimator whose input is the appropriate portion.In this paper,we also proposed an improved dichotomy to solve the ML equation.It is proved that the accuracy of the proposed method is increased,and the proposed method meets the requirements of real time.【期刊名称】《微型机与应用》【年(卷),期】2017(000)005【总页数】4页(P80-83)【关键词】线性预测编码滤波器;线性预测残差信号;最大似然估计;无偏自相关函数【作者】刘兴亮;姚剑敏;郭太良【作者单位】福州大学物理与信息工程学院,福建福州 350116;福州大学物理与信息工程学院,福建福州 350116;福州大学物理与信息工程学院,福建福州350116【正文语种】中文【中图分类】TP312混响是声音在密闭空间中经过内部障碍物等反射面多次连续反射产生的,它可使语音的质量和清晰度恶化,导致语音处理系统(电话会议、自动语音识别系统等)性能下降。
改进的小波变换HMM语音识别算法
改进的小波变换HMM语音识别算法洪淑月;施晓钟;徐皓【摘要】Recognition rate of speech recognition systems relied heavily on technology-based Hidden Markov Models-HMM model training. However the classic Baum-Welch training algorithm had a fatal flaw, namely, final solution obtained depended on the selection of the initial value, which was often only locally optimized solution. It would affect the recognition rate of the final system. To increase the recognition rate of traditional speech recognition system, it was presented an improved algorithm based on wavelet transform and HMM model. Firstly, noise in the original signal was reduced by wavelet transform, then an improved HMM model trained by speech samples and used to recognize speech. Experimental results showed that the improved algorithm, which was implemented by genetic algorithm, was practical, effective and system recognition rate was increased significantly.%语音识别系统的识别率十分依赖基于Hidden Markov Models (HMM)模型的训练技术.然而,经典的训练算法(Baum-Welch算法)有一个致命的缺陷,即所得最终解依赖于初始值的选取,只得局部最优解,这就影响了系统的最终识别率.针对传统语音识别系统识别率较低的现状,提出了一种改进的小波变换HMM语音识别算法.该算法首先通过小波变换对原始语音信号进行了降噪处理,然后使用语音样本对利用遗传算法改进后的HMM模型进行训练,并用于语音识别.实验结果表明:所提出的算法实用有效,识别率显著提高.【期刊名称】《浙江师范大学学报(自然科学版)》【年(卷),期】2011(034)004【总页数】6页(P398-403)【关键词】小波变换;降噪;HMM模型;语音识别【作者】洪淑月;施晓钟;徐皓【作者单位】浙江师范大学数理与信息工程学院,浙江金华321004;浙江师范大学行知学院,浙江金华321004;浙江师范大学数理与信息工程学院,浙江金华321004【正文语种】中文【中图分类】TP3910 引言语音识别是一个多学科交叉的领域,它与声学、语音学、语言学、数字信号处理理论、信息论、计算机科学等众多学科紧密相连[1].随着人们对语音识别认识的深入,人们对语音识别也提出了越来越高的要求.小波分析作为一种强有力的信号分析工具,近年来被广泛地应用于图像处理和语音处理中,它是时间和频率的局部变换,能有效地从信号中提取信息.通过小波变换,在信号的高频域部分,可以取得较好的时间分辨率;在信号的低频域部分,可以取得较好的频率分辨率,这种特性使得小波特别适合于语音信号处理[2].隐马尔可夫模型(Hidden Markov Models:HMM),作为语音信号的一种统计模型,目前正在语音处理各个领域中获得广泛的应用[3-4].语音识别系统的识别率十分依赖基于HMM模型的训练技术,然而经典的训练算法(Baum-Welch算法)有一个致命的缺陷,即所得最终解依赖于初始值的选取,故只得局部最优解,影响了系统的最终识别率,尤其高噪声环境下语音识别进展困难,必须寻找新的信号分析处理方法[5-6].本文改进思路,将进化算法寻找最优B初值与Baum-Welch算法相结合来训练HMM模型,使得整个语音识别系统的识别率大大提升.1 小波去噪的原理在实际运用中,去除语音信号中的背景噪声显得尤为重要.小波变换是时间和频率的局域变换,能够有效地从信号中提取信息.它不但可以检测到低信噪比信号中的边缘信号,而且可以滤去噪声从而恢复原信号.小波变换的语音降噪原理如下,令观察信号为式(1)中:有用信号噪声序列.假零均值且服从高斯分布的随机序列,即服从布.对式(1)两端作小波变换,有再令零均值、独立同分布的平稳随机信号,记u=[u(0)u(1)…u(N-1)]T,则有式(3)中表求均值运算;Q是u的协方差矩阵.令W是小波变换矩阵,对于正交小波变换,它变换,即由式(2)有令P是U的协方差矩阵,由于,因此,W是正交阵,且Q=σ2uI,所以P=σ2uI.因此,可得到一个重要的结论:平稳白噪声的正交小波变换仍然是平稳的白噪声[7].由该结论可知,对于如同式(1)的加噪声模型,经正交小波变换后,最大程度地去除了s(n)的相关性,其能量将集中在少数小波系数上.小波变换具有一种“集中”的能力,能使信号和噪声在不同尺度上所表现出的特征不同,对于信号函数,随着尺度的增大,小波变换系数也增大;对于噪声,其小波变换系数随着尺度的增大而减小.选择一个合适的阈值对小波系数进行阈值处理,就可以达到滤除噪声而保留有用信号的目的.2 HMM的改进2.1HMM 模型HMM模型作为语音信号的一种统计模型,今天正在语音处理各个领域中获得广泛的应用.语音识别系统的原理图1所示[8].HMM过程是一个双重随机过程:一重用于描述非平稳信号的短时平稳段的统计特征(信号的瞬态特征);另一重随机过程描述了每个短时平稳段如何转变到下一个短时平稳段,即短时统计特征的动态特性(隐含在观察序列中).人的言语过程本质上也是一个双重随机过程,语音信号本身是一个可观测的时变序列.可见,HMM合理地模仿了这一过程,是一种较为理想的语音信号模征参量.HMM模型通常表示成2.2HMM的3个基本问题图1 HMM语音识别系统这3个问题目前都已解决,通常情形下评估问题使用“前向-后向”算法解决,解码问题使用Viterbi算法解决,训练问题使用Baum-Welch算法解决[9].2.3 利用遗传算法改进HMM语音识别系统的识别率十分依赖基于HMM模型的训练技术,经典的训练算法(Baum-Welch算法)有一个致命的缺陷,即所得最终解依赖于初始值的选取,故往往只得局部最优解,影响了系统的最终识别率.改进思路是将遗传算法寻找最优B初值与Baum-Welch算法相结合来训练HMM模型,使得整个语音识别系统的识别率大大提升.进化Baum-Welch算法的设计如下:1)编码方案.在HMM模,参数分为A,B两部分.对于无跳跃从左向右模型,A 中有且仅有9个非零值.由于因此,A中只需5个参数形成染色体的一部分,即所以在遗传操作后还需对B部分作归一化操作.2)适应函数.遗传算法中,适应函数作为区分个体优劣的标准,需保证优秀个体的适应度比差的个体的适应度高.这里个体的适应度用各个训练样本的对数似然概率表示,即式(3)中:O(k)表示用于训练模型的第k个观测序列;P(O(k)|λ)由Viterbi算法求出.3)选择策略.文中采用了基于排名的非线性选择.在每一代中,将群体成员按适应值从高到低依次排列,按照排名分配选择概率,适应值高的个体选择概率也就相应地高.4)遗传算子和控制参数.遗传算子包含杂交算子和变异算子,它直接影响到算法的最终解.杂交算子相当于一个局部搜索操作,它产生父代附近的2个子代,而变异算子则使得个体能够跳出当前的局部搜索区域,两者的结合正好体现了进化算法的精髓所在.实验中采用了3个单点杂交,一点对应一个状态.在个体中A部分随机取一点,将2个父体该点的对应值互换;再对每一状态在B的两部分中个体随机选取一个点,将2个父体该点后的分量进行互换,这样就完成了杂交的操作.变异算子采用均匀性变异.实验中种群大小取40,杂交概率取0.7,变异概率取0.001.5)终止策略.常用的终止准则是预先设置最大进化的代数或预先设置一个适应值改善的门限值.对于前一种准则,在进化代数到达预置值时进化终止.后一种情况下,在适应值改善低于该门限值时进化停止.本系统取最大进化代数为100.3 改进型系统设计基于小波变换和改进型HMM的系统设计模型如图2所示.改进后的系统在预处理之后加入小波变换,可以对瞬间突变的语音信号进行检测与分析,有效降低原始语音信号中的噪声.小波降噪后进行端点检测,之后对语音信号进行特征参数提取MFCC,然后进行矢量量化和编码,再将编码得到的码本使用改进后的算法训练HMM,最后得到输出结果.图2 改进型系统设计框图4 实验结果分析实验基于HMM对人体语音识别系统进行.训练数据取自10人,在不同SNR(高斯白噪声)下,词汇量分别为10,20,30,40,50个,共600个实验样本,其中300个样本用于训练,另外300个用于检测实验结果.时间长度为5~10 s,采样频率为8 kHz,A/D转换精度为16 bit,并采用单声道语音进行识别测试.实验结果如表1所示.表1 4种系统的识别率比较images/BZ_130_242_405_2100_535.png系统Ⅰ 47.8 83.4 85.0 86.7 87.7 89.2系统Ⅱ 53.0 84.5 86.9 87.2 87.6 89.3系统Ⅲ 50.3 87.5 87.4 88.1 88.0 90.2系统Ⅳ 58.5 88.7 89.6 89.6 89.9 90.1 10系统Ⅰ 30.2 75.6 82.1 84.7 84.6 85.1系统Ⅱ 42.5 79.8 84.3 84.8 84.7 85.2系统Ⅲ 39.3 77.2 84.9 85.9 86.0 86.3系统Ⅳ 48.6 83.1 86.1 86.2 86.2 86.3 20系统Ⅰ 28.4 74.7 82.0 83.7 84.0 85.0系统Ⅱ 40.0 77.9 83.8 84.0 83.9 85.0系统Ⅲ 35.7 77.1 84.1 84.9 85.0 86.5系统Ⅳ46.5 82.0 85.0 85.9 86.0 86.2 30系统Ⅰ 25.4 75.0 83.3 82.1 82.5 83.0系统Ⅱ 31.7 78.5 83.4 82.3 82.4 82.7系统Ⅲ 33.3 77.3 82.3 83.1 82.9 83.1系统Ⅳ 45.2 80.4 84.0 84.5 84.4 84.0 40系统Ⅰ 23.2 72.0 79.1 80.4 80.1 81.5系统Ⅱ 30.7 76.6 83.1 80.6 80.7 81.7系统Ⅲ 29.9 76.0 82.6 81.8 83.2 82.1系统Ⅳ 44.1 80.7 83.7 84.0 83.9 84.1 50表1中,系统Ⅰ为基于HMM的语音识别系统;系统Ⅱ为基于小波变换和HMM的语音识别系统;系统Ⅲ为基于改进HMM的语音识别系统;系统Ⅳ为基于小波变换和改进HMM的语音识别系统.因此,可得到以下一些结论:1)在高噪声环境下,小波降噪对语音系统识别率可提升5% ~7%.随着语音质量(信噪比)的提高,小波降噪对识别率的改善越来越小,当信噪比大于35 dB时,小波降噪系统识别率的改善并不明显.图3是利用表1中的实验数据(词汇量为20)制成的小波降噪的识别率比较图.图3 小波变换对系统影响比较图4 系统受词汇量影响比较2)基于遗传算法的改进HMM模型对系统语音识别率有较大改善,平均提高了4个百分点,且由图4可以看出改进后的系统识别率受词汇量大小影响不大.3)改进后的语音识别系统,即系统Ⅳ在实验中表现最优,各种环境下其识别率都是最高的,基本达到了理论预期结果.5 结语提出一种语音识别系统的改进方法,通过小波变换和遗传算法对传统语音识别方法作了一定改进.改进后的语音识别算法性能提升明显,尤其是在恶劣噪声环境下,该算法基本达到了设计目的和现实要求.所提出的方法综合性能优于单独应用HMM模型和小波变换与HMM模型结合的语音识别方法.参考文献:[1]刘么和.语音识别与控制应用技术[M].北京:科学出版社,2008:1-35.[2]Zhou Dexiang,Wang Xianrong.The improvement of HMM algorithm using wavelet dek-noising in speech recognition[C]//2010 3rd International Conference on Advanced Computer Theory and Engineering(Ⅳ),Chengdu:Int Assoc Comput Sci Inf Technol,2010:4438-4441 .[3]García-Moral A I,Solera-Ureña R,Peláez-Moreno C.Data balancing for efficient training of hybrid ANN/HMM automatic speech recognition system[J].IEEE Transactions on Audio,Speech and Language Processing,2011,19:468-481.[4]Terashima R,Yoshimura T,Wakita T.Prediction method of speech recognition performance based on HMM-based speech synthesis technique[J].IEEJ Transactions on Electronics,Information and Systems,2010,130:557-564.[5]Borgstrom B J,Alwan A.HMM-based reconstruction of unreliable spectrographic data for noise robust speech recognition[J].IEEE Transactions on Audio:Speech and Language Processing,2010,18:1612-1623.[6]Hahm S J,Ohkawa Y I.Speech recognition under multiple noise environment based on multi-mixture HMM and weight optimization by the aspect model[J].IEICE Transactions on Information and Systems,2010,93(9):2407-2416.[7]胡广书.现代信号处理教程[M].北京:清华大学出版社,2004:397-398.[8]Rabiner L R,Juang B H.Fundamentals of Speech Recognition [M].New Jersey:Prentice-Hall,1999:321-370.[9]吴朝晖,杨莹春著.说话人识别模型与方法[M].北京:清华大学出版社,2009:21-76.。
consistency regularization 出处 -回复
consistency regularization 出处-回复Consistency Regularization: An Overview and ApplicationsIntroductionConsistency regularization has emerged as a powerful technique in machine learning, specifically in the field of deep learning. It aims to improve the generalization and robustness of models by encouraging consistency in their predictions. This regularization technique has found applications in various domains, including image classification, natural language processing, and speech recognition. In this article, we will provide an overview of consistency regularization, discuss its theoretical foundations, and explore its applications in different areas.Theoretical FoundationsConsistency regularization is rooted in the principle of encouraging smoothness and stability in model predictions. The underlying assumption is that small changes in the input should not significantly alter the output of a well-trained model. This principle is particularly relevant in scenarios where the training data maycontain noisy or ambiguous samples.One of the commonly used methods for achieving consistency regularization is known as consistency training. In this approach, two different input transformations are applied to the same sample, creating two augmented versions. The model is then trained to produce consistent predictions for the transformed samples. Intuitively, this process encourages the model to focus on the underlying patterns in the data rather than being influenced by specific input variations.Consistency regularization can be formulated using several loss functions. One popular choice is the mean squared error (MSE) loss, which measures the discrepancy between predictions of the original input and transformed versions. Other approaches include cross-entropy loss and Kullback-Leibler divergence.Applications in Image ClassificationConsistency regularization has yielded promising results in image classification tasks. One notable application is semi-supervised learning, where the goal is to leverage a small amount of labeleddata with a larger set of unlabeled data. By applying consistent predictions to both labeled and unlabeled data, models can effectively learn from the unlabeled data and improve their performance on the labeled data. This approach has been shown to outperform traditional supervised learning methods in scenarios with limited labeled samples.Additionally, consistency regularization has been explored in the context of adversarial attacks. Adversarial attacks attempt to fool a model by introducing subtle perturbations to the input data. By training models with consistent predictions for both original and perturbed inputs, their robustness against such attacks can be significantly improved.Applications in Natural Language ProcessingConsistency regularization has also demonstrated promising results in natural language processing (NLP) tasks. In NLP, models often face the challenge of understanding and generating coherent sentences. By applying consistency regularization, models can be trained to produce consistent predictions for different representations of the same text. This encourages the model tofocus on the meaning and semantics of the text rather than being influenced by superficial variations, such as different word order or sentence structure.Furthermore, consistency regularization can be used in machine translation tasks, where the goal is to translate text from one language to another. By enforcing consistency between translations of the same source text, models can generate more accurate and consistent translations.Applications in Speech RecognitionSpeech recognition is another domain where consistency regularization has found applications. One of the key challenges in speech recognition is handling variations in pronunciation and speaking styles. By training models with consistent predictions for different acoustic representations of the same speech utterance, models can better capture the underlying patterns and improve their accuracy in recognizing speech in different conditions. This can lead to more robust and reliable speech recognition systems in real-world scenarios.ConclusionConsistency regularization has emerged as an effective technique for improving the generalization and robustness of models in various machine learning tasks. By encouraging consistency in predictions, models can better learn the underlying patterns in the data and generalize well to unseen examples. This regularization technique has been successfully applied in image classification, natural language processing, and speech recognition tasks, among others. As research in consistency regularization continues to advance, we can expect further developments and applications in the future.。
Creating a Mexican Spanish Version of the CMU Sphinx-III Speech Recognition System
Creating a Mexican Spanish Version of the CMU Sphinx-III Speech Recognition System Armando Varela1,Heriberto Cuay´a huitl1,and Juan Arturo Nolazco-Flores21Universidad Aut´o noma de Tlaxcala,Department of Engineering and Technology,Intelligent Systems Research Group,Apartado Postal#140,90300Apizaco,Tlaxcala,Mexico.{avarela,hcuayahu}@ingenieria.uatx.mxhttp://orion.ingenieria.uatx.mx:8080/si/si.jsp2Instituto Tecnol´o gico de Estudios Superiores de Monterrey,Sucursal de Correos“J”,64849,Monterrey,Nuevo Leon,Mexico.jnolazco@itesm.mxAbstract.In this paper we present the creation of a Mexican Span-ish version of the CMU Sphinx-III speech recognition system.Wetrained acoustic and N-gram language models with a phonetic set of23phonemes.Our speech data for training and testing was collected from anauto-attendant system under telephone environments.We present exper-iments with different language models.Our best result scored an overallerror rate of6.32%.Using this version is now possible to develop speechapplications for Spanish speaking communities.This version of the CMUSphinx system is freely available for non-commercial use under request.1IntroductionToday,building a new robust Automatic Speech Recognition(ASR)system is a task of many years of effort.In the Autonomous University of Tlaxcala-Mexico, we have two goals in the ASRfield:Do research for generating a robust speech recognizer,and build speech applications for automating services.In order to achieve our goals in a short time,we had to take a baseline work.We found that the CMU(Carnegie Mellon University)Sphinx speech recognition system is freely available and currently is one of the most robust speech recognizers in English.The CMU Sphinx system enables research groups with modest budgets to quickly begin conducting research and developing applications.This arrange-ment is particularly pertinent in Latin America,where thefinancial support and experience otherwise necessary to support such research is not readily available. In the past,few research efforts have been done for Spanish and these includes work from CMU in broadcast news transcription[1,2],where basically acous-tic and language models have been trained.Our motivations for developing thiswork are due to the fact that many applications require a speech recognizer for Spanish,and because Spoken Dialogue Systems(SDS)require a robust speechrecognizer were reconfiguration and retraining is necessary.A.Sanfeliu and J.Ruiz-Shulcloper(Eds.):CIARP2003,LNCS2905,pp.251–258,2003.c Springer-Verlag Berlin Heidelberg2003252 A.Varela,H.Cuay´a huitl,and J.A.Nolazco-FloresIn this research,we have generated a lexicon and trained acoustic and lan-guage models with Mexican Spanish speech data for the CMU Sphinx speech recognition system.Our experiments are based on data collected from an auto-attendant application (CONMAT)deployed in Mexico [3],with a vocabulary of 2,288entries from names of people and places inside a university,including syn-onyms.Our speech data used for training and testing was filtered avoiding noisy utterances.Results are given in terms of the well known evaluation metric:Word Error Rate (WER).In the remainder of the paper we first provide an overview of the system in section 2.In section 3we describe the components of the Sphinx system and how these were trained.In Section 4we present experimental results.Finally,in section 5we provide our conclusions and future directions.2System OverviewThe Carnegie Mellon University Sphinx-III system is a frame-based,HMM-based,speaker-independent,continuous speech recognition system,capable of handling large vocabularies (seeFig.1).The word modeling is performed based on subword units,in terms of which all the words in the dictionary are tran-scribed.Each subword unit considered in its immediate context (triphone)is modeled by 5-state left-to-right HMM model.Data is shared across states of different triphones.These groups of HMM states sharing distributions between its member states are called senones [4].Fig.1.Architecture of the CMU Sphinx-III speech recognition system.The lexical or pronunciation model contains pronunciations for all the words of interest to the decoder.Acoustic models are based on statistical Hidden Markov models (HMMs).Sphinx-III uses a conventional backoffbigram or trigram language model.The result is a recognition hypothesis with a word lattice representing an N-best list.Creating a Mexican Spanish Version253 The feature vector computation is a two-stage process.In thefirst stage,an off-line front-end module isfirst responsible for processing the raw audio sample stream into a cepstral stream.The input is windowed,resulting in frames of du-ration25.625ms.The output is a stream of13-dimensional real-valued cepstrum vectors.The frames overlap,thus resulting in a rate of100vectors/sec.In the second stage,the stream of cepstrum vectors is converted into a feature stream. This process consists of a Cepstrum Mean-Normalization(CMN)and Automatic Gain Control(AGC)step.Thefinal speech feature vector is created by typically augmenting the cepstrum vector(after CMN and AGC)with one or more time derivatives.The feature vector in each frame is computed by concatenatingfirst and second derivatives to the cepstrum vector,giving a39-dimensional vector. 3System Components3.1LexiconThe lexicon development process consisted of defining a phonetic set and gener-ating the word pronunciations for training acoustic and language models.Table1.ASCII Phonetic Symbols for Mexican Spanish.Manner Label Example Worldbet WordPlosives p p unto p u n t ob b a˜n os b a˜n o st t ino t i n od d on de d o n d ek c asa k a s ag g an g a g a n g aFricatives f f alda f a l d as mi s mo m i s m ox j amas x a m a sAffricates tS ch ato tS a t oNasals m m ano m a n on n ada n a d a˜n b a˜n o b a˜n oSemivowels l l ado l a d oL po ll o p o L or(pe r o p e r(or pe rr o p e r ow hu eso w e s oVowels i p i so p i s oe m e sa m e s aa c a so k a s oo m o do m o d ou c u ra k u r(a254 A.Varela,H.Cuay´a huitl,and J.A.Nolazco-FloresOur approach for modeling Mexican Spanish phonetic sounds in the CMU Sphinx-III speech recognition system consisted of an adapted version from the WORLDBET Castilian Spanish phonetic set[5],which resulted in23phonemes listed in Table1.The adaptation consisted in a manual comparison of spec-trograms from words including a common phoneme;we found common sounds which we merged in ourfinal list of phonemes.The following are the modifica-tions made to the Castilian Spanish sounds set for generating a Mexican Spanish version:–Fricative/s/as in“kasa”and fricative/z/as in“mizmo”merged into/s/,–Plosive/b/as in“ba˜n os”and fricative/V/as in“aVa”merged into/b/,–Plosive/d/as in“donde”and fricative/D/as in“deDo”merged into/d/,–Plosive/g/as in“ganga”and fricative/G/as in“lago”merged into/g/,–Semi-vowels/j/as in“majo”and/L/as in“poLo”,and affricate/dZ/as in“dZugo”merged into/L/,–Nasal/n/as in“nada”and nasal/N/as in“baNko”merged into/n/,–Fricative/T/as in“luTes”was deleted due to the fact that this sound does not exist in Mexican Spanish.The vocabulary size has2,288words,which is based on names of people and places inside a university,including synonyms.The automatic generation of pronunciations was performed using a simple list of rules and exceptions. The rules determine the mapping of clusters of letters into phonemes and the exceptions list covers some words with irregular pronunciations.A Finite State Machine(FSM)was used to develop the pronunciations from the word list. 3.2Acoustic ModelsFor training acoustic models is necessary a set of featurefiles computed from the audio training data,one each for every recording in the training corpus.Each recording is transformed into a sequence of feature vectors consisting of the Mel-Frequency Cepstral Coefficients(MFCCs).The training of acoustic models is based on utterances without noise.This training was performed using3,375 utterances of speech data from an auto-attendant system,which context is names of people and places inside a university.The training process(see Fig.2)consists of the following steps:Obtain a cor-pus of training data and for each utterance,convert the audio data to a stream of feature vectors,convert the text into a sequence of linear triphone HMMs using the pronunciation lexicon,andfind the best state sequence or state align-ment through the sentence HMM for the corresponding feature vector sequence. For each senone,gather all the frames in the training corpus that mapped to that senone in the above step and build a suitable statistical model for the corresponding collection of feature vectors.The circularity in this training pro-cess is resolved using the iterative Baum-Welch or forward-backward training algorithm.Due to the fact that continuous density acoustic models are com-putationally expensive,a model is built by sub-vector quantizing the acoustic model densities(sub-vector quantizing was turned offin our work).Creating a Mexican Spanish Version255Fig.2.A block schematic diagram for training acoustic models.3.3Language ModelsThe main Language Model (LM)used by the Sphinx decoder is a conventional bigram or trigram backofflanguage model.Our LMs were constructed from the 2,288word dictionary using the CMU-Cambridge statistical language model toolkit version 2.0[6],seeFig.3.The training data consisted of 3,375transcribed utterances of speech data from an auto-attendant system.We trained bigrams and trigrams with four discounting strategies:Good Turing,Absolute,Linear,and Witten Bell.The LM probability of an entire sentence is the product of the individual word probabilities.The output from the CMU-Cambridge toolkit is an ASCII text file,and because this file can be very slow to load into memory,the LM must be compiled into a binary form.The decoder uses a disk-based LM strategy to read the binary into memory.Although the CMU-sphinx recog-nizer is capable for handling out-of-vocabulary speech,we did not set any filler models.Finally,the recognizer needs to exponenciate the LM probability using a language weight before combining the result with the acoustical likelihood.Fig.3.A block schematic diagram for training language models.256 A.Varela,H.Cuay´a huitl,and J.A.Nolazco-Flores4Experimental Results4.1Experimental SetupWe performed two experiments for evaluating the performance of the CMU Sphinx system trained with Mexican speech data(872utterances)in the con-text of an auto-attendant application:thefirst experiment considered names of people and places as independent words(i.e.any combination offirst names and last names was allowed),the second experiment considered names of people and places as only one word.Each experiment was evaluated with two different LMs.4.2Evaluation CriteriaThe evaluation of each experiment was made according to recognition accuracy and computed using the WER(Word Error Rate)metric defined by the equation 1,which align a recognized word string against the correct word string and compute the number of substitutions(S),deletions(D),and insertions(I)from the number of words in the correct sentence(N).W ER=(S+D+I)/N∗100%.(1)4.3ResultsRecognition results for each decoding stage for the CMU with Sphinx Mexican Spanish test data are shown in Tables2and3.In table2(experiment1),we can observe that the use of Good Turing discount strategy is not convenient, and the use of different n-grams does not make much difference,perhaps bigger training and test sets would yield significant differences.In the mean time,for this experiment the best option is bigrams with Witten Bell discounting strategy, but we observed problems with this approach due that this experiment can yield incorrect hypothesis,i.e.inexistent names of people and places.Thus,another solution was necessary to solve this problem.In table3(experiment2),we observe that due to the conditions of the experiment,would yield no further significant improvements with different n-grams.Despite of this,the best gains are shown in trigrams with Witten Bell discounting strategy.Table2.Word error rate in the test set after decoding from the experiment1,which considered names of people and places as independent words.Discounting Strategy Bigrams TrigramsGood Turing12.9512.88Absolute7.827.63Linear7.948.07Witten Bell7.637.75Creating a Mexican Spanish Version257 Table3.Word error rate in the test set after decoding from the experiment2,which considered names of people and places as only one word.Discounting Strategy Bigrams TrigramsGood Turing 6.88 6.44Absolute 6.38 6.38Linear 6.50 6.57Witten Bell 6.38 6.325Conclusions and Future WorkWe described the training and evaluation processes of the CMU Sphinx-III speech recognition system for Mexican Spanish.We performed two experiments in which we grouped differently the word dictionary entries.Our best results of this development considered dictionary entries as only one word for avoid-ing inexistent names of people and places inside a university.Through a simple lexicon and set of acoustic and language models,we demonstrated an accurate recognizer which scored an overall error rate of6.32%on in-vocabulary speech data.We achieved the goal of this work from which now we have a baseline product for performing research in speech recognition,which is an important component of spoken language systems.Also,with this work we can start de-velopment of speech applications with the advantage that we can retrain and adapt the recognizer according to our needs.This work was motivated due to the fact that people around the world needs to develop applications involving speech recognition for Spanish speaking communities.Therefore,the resulted lexicon,acoustic and language models are freely available for non-commercial purposes under request.An immediate future work is to provide a bridge for invoking the recog-nizer and see it as a black box,perhaps we can build a dllfile or we can pro-vide something similar as SAPI.This is indispensable for programmers who need to develop speech applications from different programming environments. Another important future direction and due that this development considers only in-vocabulary speech,we plan to retrain the recognizer considering Out-Of-Vocabulary(OOV)speech,measuring computational overhead.This is due to the fact that OOV speech is an important factor in spoken dialogue systems and degrades significantly the performance in such systems[7].Also,we plan to train Sphinx in different domains,as well as optimize configuration parameters. Finally,we plan to train Sphinx release4which was implemented in Java,and make a comparison between Sphinx III and Sphinx4in Spanish domains.All this work would be performed considering a bigger corpus. Acknowledgements.This research was possible due to the availability of the CMU Sphinx speech recognizer.We want to thank to the people involved in the258 A.Varela,H.Cuay´a huitl,and J.A.Nolazco-Floresdevelopment of the CMU Sphinx-III and of course the formers of the recognizer [8].Also,we want to thank Ben Serridge for his writing revision on this paper. References1.J.M.Huerta,E.Thayer,M.Ravishankar,and R.M.Stern:The Development ofthe1997CMU Spanish Broadcast News Transcription System.Proc.of the DARPA Broadcast News Transcription and Understanding Workshop,Landsdowne,Vir-ginia,Feb1998.2.J.M.Huerta,S.J.Chen,and R.M.Stern:The1998Carnegie Mellon UniversitySphinx-III Spanish Broadcast News Transcription System.In the proceedigns of the DARPA Broadcast News Transcription and Understanding Workshop,Herndon, Virginia,Mar1999.3.Cuay´a huitl,H.and Serridge,B.:Out-Of-Vocabulary Word Modeling and Rejectionfor Spanish Keyword Spotting Systems.Lecture Notes in Computer Science,Vol, 2313.Berlin Heidelberg New York(2002)158–167.4.Hwang,M-Y:Subphonetic Acoustic Modeling for Speaker-Independent ContinuousSpeech Recognition.Ph.D.thesis,Carnegie Mellon University,1993.5.Hieronymus L.,J.:ASCII Phonetic Symbols for World’s Languages:worldbet.Tech-nical report,Bell Labs,1993.6.P.Clarkson,and R.Rosenfeld.:Statistical Language Modeling Using the CMU-Cambridge Toolkit.In the proceedings of Eurospeech,Rodhes,Greece,1997,2707–2710.7.Farf´a n,F.,Cuay´a huitl H.,and Portilla,A.:Evaluating Dialogue Strategies in aSpoken Dialogue System for Email.In the proceedings of the IASTED Artificial Intelligence and Applications,ACTA Press,Manalm´a dena,Spain,Sep2003.8.CMU Robust Speech Group,Carnegie Mellon University./afs/cs/user/robust/www/。
科大讯飞 英文作文素材
科大讯飞英文作文素材English Answer:Introduction.In the realm of artificial intelligence, the contributions of iFLYTEK Co. Ltd. have revolutionized the landscape of voice recognition and natural language processing. iFLYTEK stands as a testament to the transformative power of innovation, setting a new standard for speech-related technologies worldwide.iFLYTEK's Core Technologies.iFLYTEK's prowess lies in its mastery of cutting-edge deep learning algorithms, vast speech data resources, and advanced algorithms. This formidable combination has enabled the company to develop a suite of core technologies that drive its industry-leading solutions.1. Speech Recognition: iFLYTEK's proprietary speech recognition engine boasts unparalleled accuracy and efficiency. It leverages deep neural networks to capture the nuanced complexities of human speech, even in noisy environments.2. Natural Language Processing: Beyond speech recognition, iFLYTEK's NLP capabilities empower machines to understand the intent and context of human language. Its advanced algorithms extract meaningful information fromtext and voice, enabling seamless communication between humans and machines.3. Machine Translation: iFLYTEK bridges linguistic barriers with its robust machine translation technology. The company's AI-powered systems translate text and speech across multiple languages, facilitating global communication and information sharing.Applications and Impact.iFLYTEK's technological advancements have foundwidespread applications in diverse industries, transforming the way we interact with technology and each other.1. Education: iFLYTEK's speech recognition technology empowers students with the ability to interact with educational materials through voice commands, enhancingtheir learning experience.2. Healthcare: iFLYTEK's NLP capabilities aid medical professionals in making informed decisions by analyzing medical records and patient data, leading to improved diagnostics and treatments.3. Customer Service: iFLYTEK's chatbot solutionsprovide businesses with automated and personalized customer support, enhancing efficiency and customer satisfaction.4. Smart Home: iFLYTEK's AI voice assistants seamlessly integrate into smart home devices, enabling users tocontrol their environment through natural language commands.Conclusion.iFLYTEK Co. Ltd. stands as a global leader in voice recognition and natural language processing, its innovative technologies revolutionizing the way we interact with machines and the world around us. From enhancing education to empowering healthcare professionals, iFLYTEK's solutions are shaping the future of AI and its impact on society.Chinese Answer:简介。
基于循环神经网络的藏语语音识别声学模型
0 引 言
藏 语 属 于汉 藏 语 系 的藏 缅 语 族 藏语 支 ,存 在 历 史悠 久 ,使用 人 口众 多r】 ],广 泛分 布 于我 国西 藏 、青 海 、甘肃 、四川 ,以及 尼 泊 尔 、印度 、巴基 斯 坦等 藏 族 聚集 地 区_3]。藏语 语 音 识 别 技 术 的发 展 ,可 有 效 解 决 藏 区与其 他地 区 之 问 的语 言 沟 通 障碍 ,促 进 民族 间交 流 ,增进 相互 了解 ,支 援藏 区经 济 、科技 、文 化等 领域 的发展 。与汉 语 、英语 等 大语种 相 比 ,藏 语 不仅
HUANG Xiaohui ,LI J ing
(1. College of Com puter Science and Technology,U niversity of Science and Technology of China, Hefei,Anhui 230027,China;2.PLA U niversity of Foreign Language,Luoyang,H enan 471003,China)
2018年 5月
JOURNAL oF CH INESE INFoRM A TIoN PRoCESSING
文 章 编 号 :1003—0077(2018)05—0049—07
May,2018
基 于循环 神 经 网络 的藏 语语 音 识别 声 学模 型
黄 晓 辉 ,李 京
(1.中 国科 学 技 术 大 学 计 算 机 科 学 与 技 术 学 院 ,安 徽 合 肥 2300471003)
型 在 保 持 同 等识 别 性 能 的 情 况 下 ,拥 有 更 高 的 训 练 和 解 码 效 率 。
关 键 词 :循 环 神 经 网络 ;藏 语 语 音 识 别 ;声 学 建 模 ;时域 卷积
毕业设计93基于连续隐马尔科夫模型的语音识别 (2)
SHANGHAI UNIVERSITY 毕业设计(论文)UNDERGRADUATE PROJECT (THESIS)论文题目基于连续隐马尔科夫模型的语音识别学院机自专业自动化学号03122669学生姓名金微指导教师李昕起讫日期2007 3.20—6.6目录摘要---------------------------------------------------------------------------2 ABSTRACT ------------------------------------------------------------------------2绪论---------------------------------------------------------------------------3第一章语音知识基础---------------------------------------------------------------6 第一节语音识别的基本内容-------------------------------------------6第二节语音识别的实现难点-------------------------------------------9第二章HMM的理论基础--------------------------------------------------------10 第一节HMM的定义----------------------------------------------------10第二节隐马尔科夫模型的数学描述---------------------------------10第三节HMM的类型----------------------------------------------------12第四节HMM的三个基本问题和解决的方-----------------------15第三章HMM算法实现的问题----------------------------------------------21 第一节HMM状态类型及参数B的选择---------------------------21第二节HMM训练时需要解决的问题-----------------------------23第四章语音识别系统的设计---------------------------------------------------32 第一节语音识别系统的开发环境-----------------------------------32第二节基于HMM的语音识别系统的设计------------------------32第三节实验结果---------------------------------------------------------49第五章结束语-------------------------------------------------------------------67致谢------------------------------------------------------------------------------68参考文献------------------------------------------------------------------------69摘要语音识别系统中最重要的部分就是声学模型的建立,隐马尔可夫模型作为语音信号的一种统计模型,由于它能够很好地描述语音信号的非平稳性和时变性,因此在语音识别领域有着广泛的应用。
HMM-based noisy speech recognition
专利名称:HMM-based noisy speech recognition发明人:Seo, Hiroshi, c/o PioneerCorporation,Komamura, Mitsuya, c/oPioneer Corporation,Toyama, Soichi, c/oPioneer Corporation申请号:EP01307875.3申请日:20010917公开号:EP1189204A3公开日:20020828专利内容由知识产权出版社提供专利附图:摘要:A multiplicative distortion Hm (cep) is subtracted from a voice HMM 5, amultiplicative distortion Ha (cep) of the uttered voice is subtracted from a noise HMM 6 formed by HMM, and the subtraction results Sm(cep) and {Nm (cep) -Ha (cep) } are combined with each other to thereby form a combined HMM 18 in the cepstrum domain.A cepstrum R^a(cep) obtained by subtracting the multiplicative distortion Ha (cep) from the cepstrum Ra(cep) of the uttered voice is compared with the distribution R^m(cep) of the combined HMM 18 in the cepstrum domain, and the combined HMM with the maximum likelihood is output as the voice recognition result.申请人:Pioneer Corporation地址:4-1 Meguro 1-chome Meguro-ku, Tokyo JP国籍:JP代理机构:Haley, Stephen更多信息请下载全文后查看。
合合信息 文本纠错:提升OCR任务准确率的方法
合合信息对于文本纠错:提升OCR任务准确率的方法理解摘要:错字率是OCR任务中的重要指标,文本纠错需要机器具备人类水平相当的语言理解能力。
随着人工智能应用的成熟,越来越多的纠错方法被提出。
近年来深度学习在OCR领域取得了巨大的成功,但OCR应用中识别错误时有出现。
错误的识别结果不仅难以阅读和理解,同时也降低文本的信息价值。
在某些领域,如医疗行业,识别错误可能带来巨大的损失。
因此如何降低OCR任务的错字率受到学术界和工业界的广泛关注。
合合信息通过本文来讲解文本纠错技术帮助更多人解决业务问题。
通常文本纠错的流程可以分为错误文本识别、候选词生成和候选词排序三个步骤。
文本纠错方法可包括基于CTC解码和使用模型两种方式,下面分别对这两种纠错方式进行介绍。
1.Beam Search该方法是针对CTC解码时的一种优化方法,这是由于当使用贪心算法进行CTC解码时忽略了一个输出可能对应多种对齐结果,导致在实际应用中错字率会颇高,并且无法与语言模型结合。
因而通过Beam Search的方法我们能够得到top最优的路径,后续也可以利用其他信息来进一步优化搜索结果。
图 1: All paths corresponding to text“a”.1.1 prefix Beam Search[1]由于有许多不同的路径在many-to-one map的过程中是相同的,当使用Beam Search 时只会选择Top N个路径,这就导致了很多有用的信息被舍弃了,如图1所示,生成“a”路径的概率为0.2·0.4+0.2·0.6+0.8·0.4=0.52,而生成blank的路径概率为0.8·0.6=0.48,当使用最佳路径的搜索方式会倾向于选择blank的路径,而不是整体概率更大的a路径。
所以就有了Prefix Beam Search,其基本思想是,在每个t时刻,计算所有当前可能输出的规整字符串(即去除连续重复和blank的字符串)的概率。
sv的知识点总结
sv的知识点总结Key Concepts of Supervised Learning:1. Data: In supervised learning, the dataset consists of input features and corresponding output labels. The input features could be any type of data, such as images, text, speech, or numerical values, while the output labels are the target variable that the algorithm is trying to predict.2. Training and Testing Data: The dataset is split into a training set and a testing set. The training set is used to train the algorithm, while the testing set is used to evaluate its performance on new, unseen data.3. Model: The model is the algorithm that is used to learn from the training data and make predictions. There are many different types of models used in supervised learning, such as linear regression, logistic regression, decision trees, support vector machines, and neural networks.4. Loss Function: The loss function is a measure of how well the model is performing. It quantifies the difference between the predicted output and the actual output, and the goal of the learning algorithm is to minimize this difference.5. Learning Algorithm: The learning algorithm is the method used to update the model's parameters in order to minimize the loss function. This typically involves using techniques such as gradient descent or backpropagation to adjust the model's weights and biases.6. Evaluation Metrics: There are several metrics used to evaluate the performance of a supervised learning model, such as accuracy, precision, recall, F1 score, and area under the ROC curve (AUC). These metrics can help determine how well the model is performing and whether it is suitable for the intended application.Key Techniques of Supervised Learning:1. Regression: Regression is a type of supervised learning in which the goal is to predict a continuous output variable. Some common regression techniques include linear regression, polynomial regression, and support vector regression.2. Classification: Classification is a type of supervised learning in which the goal is to predicta discrete output variable, such as a category or label. Some common classification techniques include logistic regression, decision trees, random forests, support vector machines, and neural networks.3. Feature Engineering: Feature engineering involves transforming the input data into a format that is more suitable for the learning algorithm. This can involve scaling, normalization, dimensionality reduction, and creating new features based on the existing data.4. Cross-Validation: Cross-validation is a technique used to assess the performance of a supervised learning model. It involves splitting the training data into multiple subsets, training the model on each subset, and evaluating its performance on the remaining data. This can help determine how well the model generalizes to new, unseen data.5. Hyperparameter Tuning: Hyperparameters are the settings of the learning algorithm that are not learned from the training data, such as the learning rate, regularization strength, and the number of hidden layers in a neural network. Hyperparameter tuning involves finding the best configuration of these settings to optimize the model's performance.Applications of Supervised Learning:1. Image Recognition: Supervised learning is used in image recognition tasks, such as identifying objects in a photo, detecting anomalies in medical images, and classifying satellite images for environmental monitoring.2. Speech Recognition: Speech recognition systems use supervised learning to transcribe spoken language into text, and to recognize spoken commands for virtual assistants and smart home devices.3. Natural Language Processing: Supervised learning is used in natural language processing tasks, such as sentiment analysis, machine translation, and named entity recognition.4. Financial Forecasting: Supervised learning is used in finance to predict stock prices, detect fraudulent transactions, and assess credit risk.5. Healthcare: Supervised learning is used in healthcare to diagnose diseases from medical imaging, predict patient outcomes, and personalize treatment plans based on patient data. Limitations of Supervised Learning:1. Data Quality: Supervised learning models are highly dependent on the quality and representativeness of the training data. If the data is biased, noisy, or incomplete, the model's predictions may be inaccurate or unreliable.2. Overfitting: Overfitting occurs when a model performs well on the training data but poorly on new, unseen data. This can happen if the model is too complex, if it is trained on too little data, or if the training data is not representative of the full range of inputs.3. Interpretability: Some supervised learning models, such as neural networks, can be difficult to interpret and understand why they make particular predictions. This can be a limitation in applications where the decision-making process needs to be transparent and explainable.4. Limited Generalization: Supervised learning models may not generalize well to new, unseen data if the training data is not diverse enough, or if the model is too specialized to the training set.In conclusion, supervised learning is a powerful and versatile approach to solving a wide range of prediction and classification problems. It has many practical applications in fields such as image recognition, speech recognition, natural language processing, finance, and healthcare. However, it also has limitations, such as its reliance on high-quality training data, the potential for overfitting, and challenges in interpreting complex models. By understanding the key concepts, techniques, and applications of supervised learning, practitioners can make informed decisions about when and how to deploy this approach to address real-world problems.。
语音识别算法中的声学建模方法总结
语音识别算法中的声学建模方法总结语音识别是一种将语音信号转化为文本的技术,广泛应用于语音助手、智能音箱、电话自动接听等各种场景中。
而在语音识别算法中,声学建模方法是其中一个关键的环节。
本文将对声学建模方法进行总结,包括高斯混合模型(Gaussian Mixture Model,GMM)、隐马尔可夫模型(Hidden Markov Model,HMM)、深度神经网络(Deep Neural Network,DNN)等方法。
首先,我们来介绍GMM方法。
GMM是一种基于统计模型的声学建模方法,它假设语音信号是由多个高斯分布组成的。
在训练过程中,我们通过最大似然估计来估计高斯分布的参数,如均值和协方差矩阵。
然后,在识别过程中,我们将输入的语音信号与每个高斯分布进行比较,选择概率最大的高斯分布作为最终的识别结果。
GMM方法常用于传统的语音识别系统中,其性能在一定程度上受到数据分布的限制。
接下来,我们介绍HMM方法。
HMM是一种基于序列建模的声学建模方法,它假设语音信号是由多个隐藏的状态序列和对应的可观测的观测序列组成的。
在训练过程中,我们通过最大似然估计来估计HMM的参数,如初始状态概率、状态转移概率和观测概率。
然后,在识别过程中,我们使用Viterbi算法来寻找最可能的状态序列,进而得到最终的识别结果。
HMM方法在语音识别中广泛应用,其优势在于对于长时序列的建模能力较好。
然而,GMM和HMM方法都存在一些问题,如GMM的参数数量较大,计算复杂度较高;HMM对于复杂的语音信号建模能力相对较弱。
因此,近年来,深度神经网络被引入到语音识别中作为一种新的声学建模方法。
深度神经网络(DNN)是一种由多层神经元构成的神经网络模型。
在语音识别中,我们可以将DNN用于声学模型的学习和预测过程中。
具体来说,我们可以将语音信号的频谱特征作为输入,通过多层的神经网络进行特征提取和模型训练,在输出层获得最终的识别结果。
相比于传统的GMM和HMM方法,DNN方法在语音识别中取得了更好的性能,其受到数据分布的限制较小,对于复杂的语音信号建模能力更强。
语音输入应用中存在的问题以及改进的方法
语音输入应用中存在的问题以及改进的方法【标题】语音输入应用中存在的问题以及改进的方法【导言】语音输入应用作为一种便捷的交互方式,正在逐渐普及和应用于日常生活中。
然而,我们也不可否认,在使用过程中,语音输入应用存在一些问题和挑战。
本文将对这些问题进行全面评估,并提出改进的方法,以期进一步提高语音输入应用的使用体验。
【正文】近年来,随着人工智能的快速发展,语音输入应用成为了人们日常生活中常见的工具之一。
然而,语音输入应用在实际使用中常常面临以下几个问题:1. 语音识别准确性有待提高语音识别技术虽然已经取得了巨大进展,但仍存在一定的准确性问题。
当语音输入涉及到方言、口音、外语等情境时,往往容易出现识别错误的情况。
对于用户而言,这可能导致输入错误或理解偏差,影响使用体验。
解决方法:a) 不断优化语音识别算法和模型,提高准确性;b) 支持多语种、方言和口音的识别,满足用户个性化需求;c) 对于识别错误的情况,提供纠正机制,让用户能够及时修正和调整。
2. 隐私和安全问题引发担忧语音输入应用存在涉及用户隐私和安全的潜在风险。
因为语音输入涉及到对用户的声音和语音数据的采集和存储,一旦这些数据被滥用或泄露,可能导致个人隐私泄露等问题,引发用户的担忧。
解决方法:a) 加强用户隐私保护,在数据采集和存储过程中遵循相关法律法规;b) 通过数据加密和安全传输等技术手段,确保语音数据的安全性;c) 提供用户选择的权利,让用户决定是否愿意共享个人数据。
3. 声音环境干扰导致使用受限在嘈杂的环境中,语音输入应用的使用往往会受到干扰,影响识别准确性和使用效果。
在咖啡厅、公共交通工具等环境中,周围的噪音可能会干扰识别系统的工作,使输入结果产生错误。
解决方法:a) 提供智能降噪功能,通过识别和过滤环境噪音,提高语音输入的识别准确性;b) 支持用户调整输入灵敏度,根据环境的不同进行灵活调整;c) 提供多种输入方式的切换,方便用户在特定环境下选择合适的交互方式。
人工智能语音识别算法实现原理解析
人工智能语音识别算法实现原理解析摘要:人工智能语音识别技术(Automatic Speech Recognition,ASR)是指通过分析和处理语音信号,将其转化为可理解的文本或命令,以实现人机交互的一种技术。
本文将介绍人工智能语音识别算法的原理及其实现过程,包括声学模型、语言模型和搜索算法。
1. 引言人工智能语音识别技术是近年来发展迅猛的一个领域,在智能手机、智能助理和语音控制等方面得到广泛应用。
其核心任务是将人类的语音信息转化为计算机能够理解和处理的文本信息,以实现自然语言与计算机语言的交互。
2. 声学模型声学模型是人工智能语音识别算法的核心组成部分。
它通过对语音信号进行建模,将语音信号与特定的语音单元(音素或子音等)进行对应。
常见的声学模型算法包括隐马尔可夫模型(Hidden Markov Model,HMM)和深度神经网络(Deep Neural Network,DNN)等。
2.1 隐马尔可夫模型隐马尔可夫模型是一种常用的声学模型算法,它假设语音信号是由一系列状态组成的序列生成的。
该模型用于描述从一个状态转移到另一个状态的概率,并且每个状态对应一个特定的语音单元。
在语音识别过程中,通过基于训练数据集的学习,确定每个语音单元与声学特征之间的对应关系,从而实现语音信号到文本的转换。
2.2 深度神经网络深度神经网络是近年来应用广泛的一种机器学习算法,也被用于语音识别的声学模型中。
它通过多层神经元的组合和连接,从输入的声学特征中提取更高层次的抽象特征,以更准确地表示语音信号。
与传统的隐马尔可夫模型相比,深度神经网络具有更好的分类性能和抗噪性。
3. 语言模型语言模型是人工智能语音识别算法的另一个关键部分。
它用于建模语音识别过程中的文本信息,以提供文本转换的先验知识和上下文背景。
常见的语言模型算法包括 n-gram 模型和循环神经网络(Recurrent Neural Network,RNN)等。
3.1 n-gram 模型n-gram 模型是一种基于统计的语言模型,它基于前文的 n-1 个词来预测下一个词的概率分布。
如何使用AI技术进行声音识别和语音合成
如何使用AI技术进行声音识别和语音合成一、声音识别1.深度学习技术2.语音识别Speech recognition is another method that can be used for sound recognition. This technology is based on natural language processing and artificial intelligence. In this method, the input is a spoken sentence and the output is the corresponding text. This technology is mainly used to recognize human speech.3.音频分析Audio analysis is also used for sound recognition. In this method, the audio is first converted into a digital signal and then analyzed to identify key features such as frequency, amplitude and envelope. These features are then used to identify the spoken words or phrases. This method is useful for identifying and recognizing speech in noisy environments.二、语音合成1.文本合成Text-to-speech is the most popular method used for voice synthesis. In this method, text is first converted into speech using natural language processing and AI algorithms. Thistechnology is widely used in virtual assistants, navigation systems and other applications.2.声音识别Voice recognition is another method that can be used for voice synthesis. In this method, the input is a spoken sentence and the output is the corresponding sound. It can be used to synthesize speech in various languages.3.融合技术。
语音识别技术的实现教程
语音识别技术的实现教程语音识别技术(Speech Recognition)是指通过计算机将语音信号转化为文字或命令的一种技术。
随着人工智能的快速发展,语音识别技术得到了广泛应用,例如智能助理、语音输入、智能家居等领域。
本文将介绍语音识别技术的实现教程,帮助读者了解语音识别技术的基本原理,以及如何实现一个简单的语音识别系统。
一、语音识别技术的基本原理语音识别技术的实现基于一系列复杂的算法和模型。
主要的基本原理包括声学模型、语言模型和搜索算法。
1. 声学模型(Acoustic Model):声学模型是语音识别的基础模型,用于将语音信号与语音单位(音素)相对应。
常用的声学模型包括隐马尔可夫模型(HMM)和深度神经网络(DNN)。
声学模型的训练需要大量的语音数据和相应的文本标记。
2. 语言模型(Language Model):语言模型用于评估语音识别系统输出结果的准确性。
它基于语音单位序列的统计规律,预测句子的概率。
常见的语言模型包括n-gram模型和循环神经网络(RNN)模型。
3. 搜索算法(Search Algorithm):搜索算法用于在候选词序列中找到最有可能的句子。
常用的搜索算法包括动态规划和维特比算法。
二、基于Python的语音识别系统实现步骤下面将介绍一个基于Python的简单语音识别系统的实现步骤,供读者参考。
1. 环境准备首先,需要在计算机上安装Python解释器和相关的依赖库。
常用的语音识别库包括SpeechRecognition、PyAudio等。
2. 录音功能使用PyAudio库进行录音功能的实现。
通过设置麦克风的参数,可以调整录音的采样率、位深度等参数。
3. 语音转文本利用SpeechRecognition库将录制的语音信号转化为文本。
SpeechRecognition库支持多种语音识别后端,例如Google、Microsoft等。
4. 文本处理对于转化后的文本,可以进行进一步的处理,例如拼写纠错、标点符号添加等。
机器人语音识别作文英语
机器人语音识别作文英语As the development of technology, speech recognition technology has been widely used in our lives. Speech recognition technology, also known as voice recognition technology, is a technology that can convert human speech into text or commands that can be recognized by machines. With the help of speech recognition technology, we can easily communicate with machines, such as smartphones, smart speakers, and robots.Speech recognition technology has greatly improved our lives. For example, when we are driving, we can use voice commands to make phone calls, send text messages, or play music without taking our hands off the steering wheel. When we are cooking, we can ask our smart speaker to play our favorite music, set a timer, or read a recipe for us. When we are watching TV, we can use our voice to change the channel, adjust the volume, or search for programs.One of the most significant applications of speechrecognition technology is in the field of robotics. Robots with speech recognition technology can understand human speech and respond accordingly. They can help us with our daily tasks, such as cleaning the house, doing the laundry, or even cooking. They can also be used in healthcare, education, and entertainment.In healthcare, robots with speech recognition technology can help doctors and nurses to take care of patients. They can remind patients to take their medicine, measure their vital signs, and provide emotional support. In education, robots with speech recognition technology can help teachers to teach students. They can answer students' questions, give feedback on their performance, and provide personalized learning experiences. In entertainment, robots with speech recognition technology can provide interactive experiences for users. They can play games, tell stories, and sing songs.However, speech recognition technology also has some limitations. For example, it may not work well in noisy environments or with people who have accents or speechimpairments. It may also have privacy concerns, as it requires access to our personal information and conversations.In conclusion, speech recognition technology has brought us many benefits and has great potential in various fields. With the continuous improvement of technology, we can expect more advanced and intelligent robots with speech recognition technology in the future. However, we should also be aware of its limitations and take measures to protect our privacy.。
基于模型自适应的声效鲁棒性语音识别算法
基于模型自适应的声效鲁棒性语音识别算法晁浩;宋成;薛霄;刘志中【摘要】Adaptation of acoustic models is presented to cope with the acoustic variability caused vocal effort variability in Mandarin speech recognition. Acoustic models trained on normal speech are applied to recognize sentences under the remaining four vocal effort modes. The maximum likelihood linear regression adaptation method is extended to the sto-chastic segment model, and the acoustic models after adaptation are used to recognize speech of corresponding vocal effort mode. Experiments conducted on“863-test”show that there is significant decrease in recognition accuracy in case of mis-matched speech models, and the recognition performance can be improved considerably by adaptation. This proves that adaptation of acoustic models is effective in solving the acoustic variability caused vocal effort.%针对声音效果变化引起的语音声学特性的改变,提出基于声学模型自适应的方法。
母语与非母语语音识别声学建模
1 概述
一个在实验室环境下工作得很好的语音识别系统在实际 应用时性能会变得不够稳健,其中一个重要的原因就是发音 的母语口音问题。以英语为例,多数中国人的英语发音并不 像母语是英语的人那样标准、清晰,而是受其母语的影响很 大,带有各种地方口音。目前语音识别技术主要基于统计模 式识别理论[1]。因此,用于模型训练的语音数据库对模型性 能会产生很大的影响。提高非母语说话人语音识别率最直接 的方法是用非母语说话人的语音训练产生识别系统,但存在 的问题是难以获得大量非母语说话人的训练语音数据,并且 用非母语发音库训练的模型会降低母语发音说话人语音识别 率。另一种方法是运用模型自适应方法,例如 MLLR 和 MAP[2],这种方法同样会在一定程度上提高非母语说话人的 语音识别率,但会降低母语为英语的说话人的语音识别率。 在实际应用场合中,一个英语识别系统最好能兼顾不同口音 的英语发音,表现出稳健性。本文先以母语为英语的大规模 语音库为基础,建立标准英语发音的语音识别模型,然后针 对小规模中国人英语发音的语音库,考虑中国人受母语影响 带来的英语发音变化,训练出能适应中国人英语发音的语音 识别模型,在这基础上采用模型融合技术,目标是保证高性 能的母语说话人的英语语音识别率,同时提高母语非英语的 说话人语音识别性能。
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Model-based Noisy Speech Recognition with Environment Parameters Estimated by Noise Adaptive Speech Recognition with Prior Kaisheng Yao ,Kuldip K.Paliwal†and Satoshi Nakamura‡Institute for Neural Computation,University of California at San Diego†School of Microelectronic Engineering,Griffith University,Australia‡ATR Spoken Language Translation Research Laboratories,Kyoto,Japankyao@ k.paliwal@.au nakamura@slt.atr.co.jpAbstractWe have proposed earlier a noise adaptive speech recognition ap-proach for recognizing speech corrupted by nonstationary noise and channel distortion.In this paper,we extend this approach. Instead of maximum likelihood estimation of environment param-eters(as done in our previous work),the present method estimates environment parameters within the Bayesian framework that is ca-pable of incorporating prior knowledge of the environment.Ex-periments are conducted on a database that contains digit utter-ances contaminated by channel distortion and nonstationary noise. Results show that this method performs better than the previous methods.1.IntroductionSpeech recognition has to be carried out often under adverse en-vironments.These environments cause distortions(mainly addi-tive background noise and channel distortion)in the speech signal. Because of this distortion,there is a mismatch between the pre-trained models and the test speech signal to be recognized.This mismatch causes degradation in speech recognition performance (the amount of degradation depends on the type and amount of distortion caused by the environment).Among many approaches for handling this mismatch problem,one common approach is to assume explicit model for representing environmental effects on speech features[1]and use this model to construct a transforma-tion which is applied either in the model space or feature space to decrease the mismatch.Though this model-based approach shows significant improvement,most of the research reported with this approach is focused on stationary noise distortion.Since the dis-tortions introduced by adverse environments are nonstationary,it is necessary to devise methods that can cope up with nonstationary distortions and improve robustness of a speech recognition system under such conditions.A number of speech recognition methods have been proposed in the literature to cope up with nonstationary environments.They can be categorized into two approaches.In thefirst approach, time-varying environment sources are modeled by hidden Markov models(HMMs)or Gaussian mixture models(GMMs)that are trained by prior measurement of environments,so that noise com-pensation is a task of identification of the underlying state/mixture sequences of the noise HMM/Mixtures,e.g.,[1].In the second approach,environment parameters are assumed to be time vary-ing and need to be estimated.We have proposed earlier a noise-adaptive speech recognition approach[2]which uses a maximum likelihood estimation method for estimating the time varying en-vironment parameters and compensates for environment effects sequentially.In this paper,we extend our work on noise-adaptive speech recognition(NASP)and estimate the environment parameters within the Bayesian pared to the previous work,which estimates environment parameters by maximum likelihood,the new method is capable of incorporating an appropriate prior knowl-edge of the environments.The outcome of this extension is a modi-fied environment parameter updating formula,incurring weighting of the estimation by sequential maximum likelihood and that from the prior.Experiments are conducted on a specifically designed database in order to test algorithm performances in nonstationary noise and channel distortion.It is shown that the proposed algo-rithm provides consistent performance improvement over previous methods for robust speech recognition.2.Model-based Noisy Speech Recognition The speech recognition problem can be described as follows.Given a set of trained modelsΛX={λxm}(whereλxmis the model of m-th speech unit trained from X)and an observation vector se-quence Y(T)=(y(1),y(2),···,y(T)),the aim is to recognize the word sequence W=(W(1),W(2),···,W(L))embedded in Y(T).Each speech unit modelλxmis aΥ-state CDHMM with state transition probability a iq(0≤a iq≤1)and each state i is modeled by a mixture of Gaussian probability density func-tions{b ik(·)}with parameter{w ik,µik,Σik}k=1,2,···,M,where M denotes the number of Gaussian mixture components in each state.µik∈R D×1andΣik∈R D×D are the mean vector and covariance matrix,respectively,of each Gaussian mixture compo-nent.D is the dimensionality of feature space.w ik is the mixture weight for state i and mixture k.In speech recognition,the modelΛX is used to decode Y(T) using the maximum a posterior(MAP)decoderˆW=arg maxWP(Y(T)|ΛX,W)PΓ(W)(1)where thefirst term is the likelihood of observation sequence Y(T) given that the word sequence is W,and the second term is denoted as the language model.2.1.Model-based Noisy Speech RecognitionIn the model-based robust speech recognition methods[1],the ef-fect of environment effects on speech feature vectors isrepresentedin terms of a model.Based on the assumption that the variances of speech,noise,and channel distortion are very small,the follow-ing non-linear transformation on the mean vectorµl ik in mixture k of state i inΛX can be used to represent environment effects onlog-spectral speech features[1][2],ˆµl ik(t)=µl ik+µl h(t)+log(1+exp(µl n(t)−µl ik(t)−µl h(t)))(2) whereµl n(t)∈R J×1andµl h(t)∈R J×1are respectively the (time-varying)mean vector for modeling statistics of the noise data{n l(t):t=1,···,T}and channel distortion{h l(t):t= 1,···,T}.Superscript l denotes log-spectral domain.We denote the parameters of the environment model,e.g.,mean vector and variance of a GMM,of the noise{n l(t):t=1,···,T}and channel distortion{h l(t):t=1,···,T}byΛN.With the estimatedΛN and certain transformation function (e.g.,Eq.(2)),Eq.(1)can be carried out asˆW=arg maxWP(Y(T)|ΛX,ΛN,W)PΓ(W)(3)This function defines the model-based noisy speech recogni-tion approach in our paper.Note that the likelihood is obtained here given speech modelΛX,word sequences W,andΛpared to Eq.(1),this approach has an extra requirement on estimation of ΛN.2.2.Environment Parameter EstimationEstimation ofΛN can be done in general using the following two approaches.Thefirst approach,e.g.,[1],assumes the dis-tortion caused by the testing environment to be stationary and es-timates HMMs/GMMs representing the testing environment.This requires environment data to train the recognition system.An-other approach[2],which is followed in this paper,treatsΛN as a time-varying model,e.g.,with a time varying mean vector,to be estimated sequentially.Our previous work estimates environ-ment parameters by maximum likelihood estimation[2].In this paper,we extend it to environment parameter estimation within the Bayesian framework.Denote the estimated environment parameter sequence till frame t−1asΛN(t−1)=(ˆλN(1),ˆλN(2),···,ˆλN(t−1)),whereˆλN (t−1)is the parameter estimated in the previous frame.IfλN(t),which is assumed to be random vector taking values in R J,is the parameter vector to be estimated from the sequence Y(t)=(y(1),y(2),···,y(t))till frame t with probability den-sity function(p.d.f.)P(Y(t)|ΛX,(ΛN(t−1),λN(t))),then the Bayesian estimation,in particular,the maximum a posterior prob-ability(MAP)estimation,ˆλMAPN (t),is defined as the mode of theposterior p.d.f.ofλN(t)denoted as P(λN(t)|Y(t),ΛX),i.e.,ˆλMAP N (t)=arg maxλN(t)P(λN(t)|Y(t),ΛX)(4)=arg maxλN(t)P(Y(t)|ΛX,(ΛN(t−1),λN(t)))P(λN(t))where the second term in the right side of equation is the prior density ofλN(t).Note that there is hidden state sequence S(t)in the above like-lihood function P(Y(t)|ΛX,(ΛN(t−1),λN(t))).In fact,some previous works,e.g.,[2],provide EM type recursive estimation pro-cedures for maximizing this likelihood function.It follows that in the context of hidden state sequence S(t)in HMM,the same iter-ative procedure can be used to estimate the mode of the posterior density by appending a new cost function from prior density to that used for maximizing the likelihood function[3].In particu-lar,an objective function based on sequential Kullback proximal algorithm[2]is modified toˆλMAPN(t)=arg maxλN(t)Q t(ˆλN(t−1);λN(t))(5)−(βt−1)I(ˆλN(t−1);λN(t))+log P(λN(t))where the auxiliary function Q t(·)is defined asQ t(ˆλN(t−1);λN(t))=S(t)P(S(t)|Y(t),ΛX,(ΛN(t−1),ˆλN(t−1))) log{P(Y(t),S(t)|ΛX,(ΛN(t−1),λN(t)))}(6) In Eq.(5),βt∈R+works as a relaxation factor,and the Kullback-Leibler(K-L)divergence,I(ˆλN(t−1);λN(t))is given as,I(ˆλN(t−1);λN(t))=(7)S(t)P(S(t)|Y(t),ΛX,(ΛN(t−1),ˆλN(t−1))) logP(S(t)|Y(t),ΛX,(ΛN(t−1),ˆλN(t−1)))P(S(t)|Y(t),ΛX,(ΛN(t−1),λN(t))) Note that thefirst two terms in Eq.(5)are for maximizing the likelihood.It is known that ifλN(t)does not have informative prior,Eq.(5)is the same as maximum likelihood estimation.If it is assumed that the prior density ofλN(t)is a Gaussian,i.e., P(λN(t))=N(λN(t);λ0N,Σ0N),then it can be seen that the parameter estimates are a weighted sum of the prior parameters and that from the observed data sequence.Now,by second order expansion of Eq.(5),parameter updating can be similarly devised as that in[2],i.e.,ˆλMAPN(t)←ˆλN(t−1)−βt∂2Q t(ˆλN(t−1);λN(t))∂λN(t)+ (1−βt)∂2l t(λN(t))∂λN(t)2+∂2log P(λN(t))∂λN(t)2−1(8)∂Qt(ˆλN(t−1);λN(t))∂λN(t)+∂log P(λN(t))∂λN(t)λN(t)=ˆλN(t−1)where thefirst-and second-order derivations of the auxiliary func-tion,∂Q t(ˆλN(t−1);λN(t))∂λN(t)and∂2Q t(ˆλN(t−1);λN(t))∂λN(t)2,are respec-tively given in Eq.(12)and Eq.(13)in[2].The second-orderderivation of the log-likelihood,∂2l t(λN(t))∂λN(t)2,is given in Eq.(14) in[2].Thefirst-and second-order derivative of the log prior density function are respectively given as,∂log P(λN(t))∂λN(t)=−(Σ0N)−1(λN(t)−λ0N)(9)∂2log P(λN(t))∂λN(t)2=−(Σ0N)−1(10)Compared to Eq.(11)in[2],it is seen that,the cost from prior density adds a constant matrix of Eq.(10),and a weighted vector of Eq.(9)into the second-and thefirst-order derivative of the objective function,respectively.TheΣ0N is a hyper-parameter, which controls the relative importance of the prior to theestimatefrom maximum likelihood.E.g.,when the elements in the matrix Σ0N approaches infinity,the prior does not have contribution to the estimation in Eq.(8).On the contrary,when the elements inΣ0N approaches to0,the estimation by Eq.(8)is in factλ0N.Once theλMAPN (t)is obtained,it substitutesˆλN(t−1)forupdating at the next frame by Eq.(8).2.2.1.Derivation of Time-varying Channel Parameter Estimation -A Particular CaseDue to limited space,we only outline environment parameter es-timation for channel distortions1.In this case,the environment model isλN(t)=µl h(t).The model of environment effects shown in Eq.(2)relatesλN(t)to the log-likelihood function of observa-tion y(t)given state i,mixture k,and the modelλN(t)bylog b ik(y(t))=−D2log(2π)(11)−1 2log|Σik|−12(y(t)−ˆµik(t))TΣ−1ik(y(t)−ˆµik(t))where superscript T denotes transpose operation.By differentiation of the log-likelihood function w.r.t.the en-vironment parameter,we see the“contribution”of the environment parameter to the change of the log-likelihood,i.e.,∂log b ik(y(t))∂λN(t)=GˆλN∂ˆµl ik(t)∂λN(t)(12)where the jj th element in diagonal matrices GˆλN∈R J×J isgiven as GˆλN jj =Dd=1[z d j(y t(d)−ˆµikd(t−1))Σ2ikd].z d j is the DCTcoefficient.Thefirst-order differential term,∂ˆµl ik(t)∂λN(t),in Eq.(12)is ob-tained by differentiation Eq.(2)w.r.t.µl h(t),and,for each element µl hj(t)in the environment parameterλl N(t),it is given as∂ˆµl ik(t)hj =1−exp(µl nj(t)−µlikj−µl hj(t))nj ikj hj(13)Using chain rule,this derivative of log-likelihood w.r.t.the channel distortion parameters contributes to∂Q t(ˆλN(t−1);λN(t))∂λN(t)and∂2l t(λN(t))∂λN(t)2through Eq.(12)and Eq.(14)in[2],respectively.Suppose that the informative prior on channel is available,the Bayesian updating in Eq.(8)then combines the updating from the above derivatives of log-likelihood w.r.t.to the channel distortion parameterµl h(t)and that from the prior.2Furthermore,even a coarse choice of the hyper-parameterΣ0N may still benefit the updating,since adding(Σ0N)−1into the inverse matrix in Eq.(8) reduces the condition number of this inverse matrix,which results in a more stable estimation than that without adding the(Σ0N)−1.1Please refer to[4]for detailed updating formulae.2Remark:In this paper,a model of environment effects in Eq.(2)is assumed for log-spectral speech features.However,the above environment model estimation procedure is not limited to this particular environment ef-fects model.The procedure is general and,given a correct model reflecting environment effects on other speech features,e.g.,LDA based features,the procedure can be possibly utilized to these speech features.3.Experimental ResultsA database was designed to evaluate system performances in non-stationary noise and channel distortion.Training set consisted of 8840clean utterances from Aurora2database.Testing set in-cluded1000utterances in each of four SNR conditions.These testing utterances were generated from testing utterances in Au-rora2database by convolution with a50-tap FIRfilter(simulating the channel distortion)and corruption by simulated non-stationary noise with white spectral characteristics.The spectral shape of the distortionfilter can be seen in Fig.1,and the spectral evo-lution of the non-stationary noise can be seen in Fig. 2.The signal-to-noise ratio(SNR)in the degraded speech was measured by SNR=10log10energy of filtered speech.The SNRs were 20.5dB,15.6dB,11.1dB,and6.8dB.The speech recognizer was based on whole-word HMM.Each digit was modeled by18states,and each state has3diagonal Gaus-sian mixture densities.Afilter-bank with twenty-sixfilters was used in the binning stage.The window size was25ms and time-shift was10ms.Features were MFCC plus C0,and their∆and ∆∆coefficients that,as a whole,had39dimension.The prior environment parameters in Eq.(9)and Eq.(10) were chosen asλ0N=[µl n(0)T0T]T.The variance-covariance matrixΣ0N=I2J×2J.The relaxation factorβt in Eq.(8)was set to0.8.Initialization of the noise parameterµl n(t)was made by setting it to be the mean vector of silence segments in the testing environments.µl h(t)was initialized to be a zero vector.3.1.Estimation of Channel and Noise ParametersWe show the performance of our method for the estimation of chan-nel and noise parameters through an example.In this example,we use all the utterance in the testing set at11.1dB SNR.In Fig.1,the estimated channel responses(log squared-magnitude)are shown at the initialization,the end offirst utterance,the end of second utterance,the end of third utterance,and the end of the testing set, and are compared with the response of true channel.Note that the estimation is carried out in each Melfilter-bank pared to the true channel response,the initialization does not provide shape of the true channel response,which is bent in low-and high frequency.In thefigure,“the1st utterance”means the estimate of channel response at the end of thefirst utterance,“the2nd ut-terance”means the estimate of channel response at the end of the second utterance,and so on.It can be seen from thisfigure that our noise-adaptive speech recognition approach provides updated channel response estimates that follow the general spectral slope of the channel.As the number of testing utterances increase,the slope of the estimated channel responses is closer to the true chan-nel response.However,in the high frequency end,the estimates are not as sharply bent as the true channel response.This may be attributed to the lack of speech energy at the high frequency end.In Fig.2,the estimated noise spectrum is shown in the7th Mel filter-bank bin as well as the true evolution of the noise spectrum along the time in the bin.The noise spectrum estimate is seen to evolve from poor initialization to the true noise spectrum.Note that the true noise power changes its value along the time with increasing frequency.Below a certain changing rate of the true noise power,the noise adaptive speech recognition provides the noise spectrum estimates that can follow the evolution of the true noise spectrum.Although rapid change of the true noise power spectrum makes the estimation more difficult to follow the trend, the estimated noise parameter is still within the range ofnoisespectrum evolution.Figure 1:Estimated channel responses at 11.1dB.Figure 2:Estimated noise spectra at 7th Mel filter bank at 11.1dB.3.2.Speech Recognition ResultsWe report the speech recognition results for the distorted testing utterances under different SNR conditions.The word recognition accuracy on the clean test set was 99.6%.A system with the pro-posed noise adaptive speech recognition,denoted as NASP,with a prior environment model defined before was compared to the following systems:1)Baseline:Recognized degraded speech di-rectly,2)Parallel model combination assuming stationary noise,denoted as SNA:Obtained noise parameter estimation from whole testing utterances and applied Log-Add noise compensation [1]to adapt acoustic models,3)Cepstral mean normalization,denoted as CMN,and 4)Speech enhancement using Wiener filtering 3,de-noted as ENH:A Wiener filter is applied for enhancement of testing utterances before speech feature extraction.The recognition results are summarized in Table 1.The base-line performance dropped rapidly as the SNR decreased from 20.5dB to 6.7dB.It is found that the system ENH,which uses Wiener fil-ter to enhance signals,had poor performances in this database.Also,the SNA system did not improve system performances over Baseline at all.Since both ENH and SNA systems estimated noise parameters from silence segments,in this particular task,the esti-mated noise parameters may not represent accurately the true noise statistics in speech utterances.The CMN system improved recog-nition accuracies over baseline when SNRs are 15.6dB or below,but did not get improvement in higher SNR conditions.The NASP system jointly compensated channel distortion and additive time-varying noise,and its performance was consistently among the best in the evaluated systems.3The Wienerfilter was implemented according to proposal [5].Table 1:Word Accuracy (in %)in the nonstationary environments achieved by the noise adaptive system (denoted as NASP)with βt =0.8,ρ=0.95in comparison with Baseline (recognized degraded speech directly),speech enhancement by Wiener filter (denoted as ENH),Log-Add [1]noise compensation assuming sta-tionary noise (denoted as SNA),and the system employing cepstral mean normalization (denoted CMN).SNR (dB) 6.711.115.620.5Baseline 73.382.289.995.3ENH 78.981.884.185.9SNA 73.682.189.294.7CMN 76.185.491.494.1NASP77.390.293.797.2It is worth noting the comparison between baseline and CMN.For 15.6dB SNR and below,compared to the baseline,CMN provided robustness in the sense that it compensated channel dis-tortions to some extent.At 20.5dB SNR,although the additive noise energy in average was relatively small (in this situation,the environment effects is usually considered as being dominated by channel distortions),the environment could not be considered as stationary due to fluctuation of additive noise powers as shown in Fig.2.Thus,it is appropriate to consider compensating channel distortions and additive noise jointly and dynamically.4.SummaryWe have proposed a noise adaptive speech recognition approach for recognizing speech corrupted by nonstationary noise and channel distortion.Instead of maximum likelihood estimation of environ-ment parameters (as done in our previous work),the present method estimates environment parameters within the Bayesian framework that is capable of incorporating prior knowledge of the environ-ment.We have conducted on a database that contains digit utter-ances contaminated by channel distortion and nonstationary noise.Results show that this method performs better than the other meth-ods.We plan to extend this research work by incorporating other schemes to construct informative priors and other models of envi-ronment effects.5.References[1]M.J.F.Gales and S.J.Young,“Robust speech recognition in additiveand convolutional noise using parallel model combination,”Computer Speech and Language ,vol.9,pp.289–307,1995.[2]K.Yao,K.Paliwal,and S.Nakamura,“Noise adaptive speech recog-nition in time-varying noise based on sequential Kullback proximal algorithm,”in ICASSP ,2002,pp.189–192.[3]J.-L.Gauvain and C.-H.Lee,“Maximum a posteriori estimation formultivariate Gaussian mixture observations of Markov chains,”IEEE Trans.on Speech and Audio Processing ,vol.2,no.2,pp.291–298,1994.[4]K.Yao,K.Paliwal,and S.Nakamura,“A noise adaptive speech recog-nition approach to robust speech recognition in time-varying environ-ments,”Tech.Rep.TR-SLT-0016,ATR SLT,2002.[5]ETSI,“Speech processing,transmission and quality aspects (STQ);distributed speech recognition;advanced front-end feature extraction algorithm;compression algorithms,”Tech.Rep.ETSI ES 202050,ETSI,2002.。