有效频带多分辨率特征提取及说话人年龄识别

针对文本无关非特定说话人年龄识别,本文提出了一种基于有效频带多分辨率特征的统计分析识别方法。输入语音,通过小波包变换进行有效频带分解,然后将各有效频带的小波包系数连接构成一个整体计算美尔频率倒谱系数,得到有效频带多分辨率特征参数WPMFC（Wavelet Packet Mel-Frequency Cepstrum）,说话人按年龄划分为儿童、青年、中年和老年四个阶段,并进一步按性别训练各年龄段语音得到8个高斯混合模型。测试语音依据最大似然准则进行识别判决。实验对本文提出的方法与传统的短时谱统计分析方法进行了比较,结果显示本文提出的方法有较好的识别性能,集内平均识别率达到65.17%。同时,实验结果也说明相对语音文本变化的影响,不同说话人发音特征的变化对识别性能的影响更大。      

A low-power integrated circuit for remote speech recognition

In this paper, a low-power, low-voltage speech processing system is presented. The system is intended to he used in remote speech recognition applications where feature extraction is performed on terminal and high-complexity recognition tasks and moved to a remote server accessed through a radio link. The proposed system is based on a CMOS feature extraction chip for speech recognition that computes 15 cepstrum parameters, each 8 ms, and dissipates 30 μW at 0.9-V supply. Single-cell battery operation is achieved. Processing relies on a novel feature extraction algorithm using 1-bit A/D conversion of the input speech signal. The chip has been implemented as a gate array in a standard 0.5-μm, three-metal CMOS technology. The average energy required to process a single word of the TI46 speech corpus is 10 μJ. It achieves recognition rates over 98% in isolated-word speech recognition tasks     

基于GARCH模型的改进的倒谱域特征参数补偿算法

本文提出了一种改进的倒谱域特征参数补偿算法GMCSM。根据语音信号的时变特性,GMCSM算法使用广义自回归条件异方差(Generalized Auto-Regressive Conditional Heteroscedasticity,GARCH)模型对语音信号的方差进行建模。实验数据表明,与常规倒谱相减法CSM和MEMCSM相比,GMCSM能够更有效地补偿因加性噪声引起的倒谱特征参数失真,减少识别的错误率,特别是在信噪比较低的情况下,GMCSM的性能更为显著。     

Selecting Good Speech Features for Recognition

This paper describes a method to select a suitable feature for speech recognition using information theoretic measure. Conventional speech recognition systems heuristically choose a portion of frequency components, cepstrum, mel-cepstrum, energy, and their time differences of speech waveforms as their speech features. However, these systems never have good performance if the selected features are not suitable for speech recognition. Since the recognition rate is the only performance measure of speech recognition system, it is hard to judge how suitable the selected feature is. To solve this problem, it is essential to analyze the feature itself, and measure how good the feature itself is. Good speech features should contain all of the class-related information and as small amount of the class-irrelevant variation as possible. In this paper, we suggest a method to measure the class-related information and the amount of the class-irrelevant variation based on the Shannon's information theory. Using this method, we compare the mel-scaled FFT, cepstrum, mel-cepstrum, and wavelet features of the TIMIT speech data. The result shows that, among these features, the mel-scaled FFT is the best feature for speech recognition based on the proposed measure.     

一种基于非线性特征的应力影响下变异语音识别方法

考虑到变异语音产生的非线性特点,本文提出了一种基于TEO能量算子倒谱特征的应力影响下变异语音识别方法。先将语音信号分割成21个不同频带的信号,然后计算TEO能量,最后进行对数运算和离散余弦变换。对航空模拟飞行器中采集的小词表特定人的识别实验,采用非线性分析的基于TEO能量算子倒谱特征的方法,能有效地提高变异语音的识别性能,比传统的基于MFCC特征的方法识别率提高了11.3％。     

基于TMS320C54XDSP的语音识别装置的研究与实践

运用TMS320C5416实现了语音自动识别装置。该装置利用一种新的语音信号r阶的倒谱线性回归系数等参数构成识别的特征矢量集，运用模糊矢量量化技术实现了特定人的语音识别。实验结果表明该系统具有识别精度高、识别速度快等特点．是一种语音自动识别装置的有效的硬件实现方案。     

数字助听器中单通道语音增强算法的研究

传统多通道补偿算法忽略保护语音特征,容易造成语音结构变形和识别率低等问题。为了解决上述问题,本文提出了一种基于多分辨率小波的单通道语音增强算法。利用多分辨率小波对信号进行分解与重构,提取出语音信号的频谱包络,得到特征点出其信息并依此计算补偿增益,再利用插值算法计算出整个频谱的增益并对其进行响度补偿。仿真实验以及主观性能测试的各项结果均表明该算法能够在对语音进行补偿的同时有效地保护语音特征,提高言语识别率,达到比较理想的效果。      

基于归一化自相关的语音基频特征提取

语音的基频（也称音高、基音周期或F0）及其变化规律是语音信号的一个重要特征,在语音情绪识别、声纹识别中有重要的应用。而语音基频的提取一直是语音信号处理中的难点,这也是语音基频特征未能广泛应用于语音识别等应用的重要原因,因此准确高效的提取音高在语音信号处理中能够有重要的意义。本文基于归一化自相关函数,结合倒频谱方法,提取了一种改进的基于归一化自相关的语音基频提取算法,实验证明该方法在基频提取中取得了较好的结果。     

基于最小相位分解的语音去混响

为解决语音通信中的混响干扰问题,提出了一种基于最小相位分解的,可用于单通道语音的去混响方法。根据信号最小相位分解的原理,将接收到的含噪带混响的语音信号分解成最小相位部分和全通部分,对其中的最小相位部分进行复倒谱域的滤波处理,再与全通部分进行合成以实现混响的去除。通过仿真实验来实现这种方法,并与另一种改进过的复倒谱域滤波的去混响方法进行比较,实验结果表明这种方法相对较好。     

语音信号去混响原理与技术

语音信号去混响技术在通信、语言识别等方面有重要应用。介绍了国内外相关研究动态和方法,阐述了声音混响过程和倒谱法去混响原理,简要介绍了传声器阵列-倒谱法去混响技术。     

一种改进的语音信号特征参数提取算法研究

为了提高语音信号的识别率,提出了一种改进的语音信号特征提取算法。该算法在MFCC参数的基础上,增加每帧信号的短时能量和短时过零率,使得新参数能够更为准确地表征语音信号。通过仿真实验。说明了新特征参数取得了较高的识别率。     

Performance Evaluation of Silence-Feature Normalization Model using Cepstrum Features of Noise Signals

Speech enhancement algorithms play an important role in speech signal processing. Over the past several decades, many algorithms have been studied for speech enhancement. A speech enhancement algorithm uses a noise removal method and a statistical model filter to analyze the speech signal in the frequency domain. Spectral subtraction and Wiener filters have been used as representative algorithms. These algorithms have excellent speech enhancement performance, but suffer from deterioration in performance due to specific noise or low signal-to-noise ratio (SNR) environments. In addition, according to estimations of erroneous noise, a noise existing in a voice signal is maintained so that a spectrum corresponding to a voice signal is distorted, or a frame corresponding to a voice signal cannot be retrieved, and voice recognition performance deteriorates. The problem of deterioration in speech recognition performance arises from the difference between speech recognition and training model. We use silence-feature normalization model as a methodology to improve the recognition rate resulting from the difference in the noisy environments. Conventional silence-feature normalization has a problem in that the silent part of the energy increases, which affects recognition performance due to unclear boundaries categorizing the voice. In this study, we use the cepstrum feature of the noise signals in the silence-feature normalization model to improve the performance of silence-feature normalization in a signal with a low SNR by setting a reference value for voiced and unvoiced classification. As a result of recognition rate confirmation, the recognition rates improve in performance, compared with other methods.     

一种基于短时倒谱速变率的语音信号平滑端点检测方法

短波通信以其天波传播特性,在通信领域具有其他通信手段无法替代的地位。为解决短波通信中信噪比较低,噪声信号严重影响语音处理效果的问题,提出了一种基于短时倒谱速变率的平滑端点检测方法,通过检测噪声信号的倒谱特征,初步确定语音信号端点,然后加入平滑优化处理以及Holdon设计,降低由于倒谱突变造成的误判。实测信号仿真实验证明,该方法在不过多加重系统负担的前提下,取得了良好的效果。     

一种带噪语音信号端点检测方法研究

端点检测是语音识别中的一个重要环节.当信噪比较低时,传统的端点检测方法不能有效的工作,影响了系统的识别率.为此,本文提出了一种更有效的端点检测算法--基于LPC美尔倒谱特征的端点检测方法,它是基于LPC距离方法的一种改进.实验证明,该算法在低信噪比的情况下,能够准确的检测出语音信号.通过对三种不同的端点检测算法的比较,证明了基于LPC美尔倒谱特征算法的检测正确率较高.     

Bayesian channel equalisation and robust features for speechrecognition

The use of a speech recognition system with telephone channel environments, or different microphones, requires channel equalisation. In speech recognition, the speech model provides a bank of statistical information that can be used in the channel identification and equalisation process. The authors consider HMM-based channel equalisation, and present results demonstrating that substantial improvement can be obtained through the equalisation process. An alternative method, for speech recognition, is to use a feature set which is more robust to channel distortion. Channel distortions result in an amplitude tilt of the speech cepstrum, and therefore differential cepstral features provide a measure of immunity to channel distortions. In particular the cepstral-time feature matrix, in addition to providing a framework for representing speech dynamics, can be made robust to channel distortions. The authors present results demonstrating that a major advantage of cepstral-time matrices is their channel insensitive character     

基于线性预测残差倒谱的多语音基音频率检测

本文提出了一种基于线性预测残差倒谱的多语音基音频率检测算法,该算法首先对混合语音信号进行线性预测分析,进而计算预测信号与原混合信号的残差,并对残差信号做倒谱变换,得到混合语音信号的线性预测残差倒谱;然后在该信号的残差倒谱中,结合图像处理的技术,利用语音信号基音倒频匹配法检测出多语音信号的基音频率;最后在基音标定的过程中,本文算法利用语音信号的连续特性,依据信号基音频率前后差距变化最小原则标记出各基音所属话者。实验结果表明,本文提出的算法在弱回声及无回声的情况下能快速有效地从单声道混合语音信号中检测出多语音基音信息。     

模糊C-均值(FCM)聚类法与矢量量化法相结合用于说话人识别

该文提出了一种将模糊C-均值聚类法与矢量量化法相结合进行说话人识别的方法。该算法将从语音信号中提取的 12阶 LPC(线性预测编码)倒谱系数作为待分类样本的 12个指标,先用矢量量化法求出每个说话人表征特征参数的码书,作为模糊聚类算法的聚类中心,最后将待识别的特征矢量以得到的码书为聚类中心,进行聚类识别。该算法所使用的特征参数较少,计算比较简单,但识别率较矢量量化法高。     

一种孤立词语音识别的实现方法及改进

通过对特征提取模块2个重要部分：端点检测和线性预测倒谱（LPCC）相关原理的介绍分析,阐述了一种以线性预测倒谱（LPCC）为基础,进行特征提取的孤立词语音识别的具体实现方法,并对该方法所描述的系统进行了软件建模。通过分析研究,给出了提高识别率的具体改进方案。最后使用Matlab软件对相关方法及结论进行了验证,表明该方法确实在传统方法的基础上提高了识别率,且速度较快,具有实用性和良好的硬件可移植性,并讨论了它在一些关键环节的未来实现及改进方向。     

Feature classification criterion for missing features mask estimation in robust speaker recognition

Currently, many speaker recognition applications must handle speech corrupted by environmental additive noise without having a priori knowledge about the characteristics of noise. Some previous works in speaker recognition have used the missing feature (MF) approach to compensate for noise. In most of those applications, the spectral reliability decision step is performed using the signal to noise ratio (SNR) criterion, which attempts to directly measure the relative signal to noise energy at each frequency. An alternative approach to spectral data reliability has been used with some success in the MF approach to speech recognition. Here, we compare the use of this new criterion with the SNR criterion for MF mask estimation in speaker recognition. The new reliability decision is based on the extraction and analysis of several spectro-temporal features from across the entire speech frame, but not across the time, which highlight the differences between spectral regions dominated by speech and by noise. We call it the feature classification (FC) criterion. It uses several spectral features to establish spectrogram reliability unlike SNR criterion that relies only in one feature: SNR. We evaluated our proposal through speaker verification experiments, in Ahumada speech database corrupted by different types of noise at various SNR levels. Experiments demonstrated that the FC criterion achieves considerably better recognition accuracy than the SNR criterion in the speaker verification tasks tested.     

抗噪声的小波谱压缩特征提取算法在语音识别中的应用

针对语音识别实际应用过程中的噪声问题,给出了一种新的抗噪声的特征提取算法,即先利用小波变换将语音信号进行小波子带分解,再根据人耳的听觉掩蔽效应,由谱压缩的技术,将小波变换后的子带语音信号进行压缩,从而提取其对应的语音特征。通过MATLAB软件建立实验平台,仿真实验结果表明该语音特征可以在噪声环境下得到较高的识别率。新的特征参数即充分利用了小波的抗噪声特性又有效地降低了语音识别中的训练环境和识别环境间的失配,具有抗噪声的特点。