语音人机界面标引系统的实现

本文实现了一个媒体资产管理中的语音人机界面标引系统。系统以连续混合高斯隐马尔可夫模型为基础，采用分层构筑维特比算法进行训练和识别。为实现标引的实时性．采用实时计算的方法。为了减少计算量．并没有将状态持续时间分布引入Viterbi译码，而是将其作为后处理部分。对于数字识别，以声调作为辅助判决。以此做了一个体育赛事的词库．测试表明，标引系统首选识别率达到93．5％，前五选识别率达到98％。     

基于HMM/SVM的抗噪语音特征提取及优化

为了提高语音识别的鲁棒性,提出一种新的特征组合方法。方法基于F比对梅尔频率倒谱系数(MFCC)进行加权优化,同时将不同特征组合输入到语音隐马尔科夫模型(HMM)进行训练,得到具有抗噪性的最佳组合,并采用主成分分析(PCA)进行降维,增加支持向量机(SVM)分类器作为后处理器。实验表明,改进的MFCC、短时平均能量和Teager能量算子组合参数识别效果最优,识别率达到90. 48%。PCA降维后识别率降低了0. 4%,提升了计算速度。增加后处理器,系统识别率达到95. 25%,提高了系统的识别效率和分类决策力,相对于常规识别方法,准确率有所提高。     

属性嵌套计算网格智能算法实现车牌汉字精准识别的设计

设计了环境较差情况下高效精准、辨识汉字的智能车牌识别算法,通过引进属性嵌套计算网格实现了汉字高效辨识的车牌识别算法;算法应用结果表明:算法设计的网格密度与识别率是成正比的,采用的属性嵌套计算网格模型,显著地改进了字符的识别率;将属性计算网格算法与属性嵌套计算网格算法对比可知,采用属性嵌套计算网格算法识别率是98.7%,识别率明显较高;设计算法系统不仅实现了汉字识别的稳定、智能特性,同时表现了抵抗较强外界干扰的特性,这一研究对于智能化汉字识别有明显的借鉴价值。     

面向互动电视的手势交互系统

针对用户与互动电视交互的需求,设计实现一种基于智能移动终端加速度传感器的手势交互系统。考虑到智能移动设备资源及计算能力有限的特点,采用简单高效的时域特征提取方法,对加速度信号进行平稳降噪、去冗余和归一化处理,并用SVM进行分类和识别。手势识别结果应用于基于Android平台的机顶盒原型系统,实现用户与电视的实时交互。实验结果表明该系统实现了电视常用手势的准确识别,识别率达到了96%,具有一定的实用价值。     

集成汉英OCR系统识别中文名片

汉英双语混排识别是构造中文自动文档图像处理系统时常会遇到的一个问题。只有采用一种有效的方法集成现有汉英识别引擎,才可能高质量地识别混排文档。该文应用适当干预和多层次语言判断的汉英OCR系统集成原则,集成OCR系统识别中文名片。实验数据表明,利用该原则构造的系统确实能有效集成汉英识别引擎,在纯中文识别率为89.86%,纯英文识别率为91.20%的情况下,使名片最终总体识别率达到了93.45%,较好地解决了汉英混排名片的识别问题。     

雷达对抗侦察装备自动测试中的界面识别研究

在对雷达对抗侦察装备进行整体性能自动测试时,是通过自动识别装备内部计算机界面的方法来实现的;采用BP神经网络算法可以得到很好的识别率;首先介绍了BP神经网络的识别原理,其次给出了系统的总体设计思路,对系统服务器和客户端进行了设计,并给出了识别模块的实现;最后针对装备界面显示的特点,对识别系统进行了改进,有效提高了识别率;整个系统速度快、识别效率高、准确性高,经实际测试,识别系统对于数字的识别成功率达到了98%,而对英文字母的识别成功率达到了96.5%,为装备的自动测试打下了很好的基础,具有一定的应用前景。     

基于小波熵的辐射源指纹特征提取方法

在对辐射源信号进行小波分析的基础上,提出一种基于小波熵的辐射源指纹特征提取方法。 首先计算辐射源信号的功率谱,对功率谱进行连续小波变换,提取不同尺度下小波系数的熵 特征作为辐射源信号指纹特征。识别分类器采用概率神经网络,对20部手持机进行识别实验 ,并与传统矩形积分双谱进行对比。实验结果表明,该方法能够把辐射源信号的时频特性信 息通过小波系数的熵特征映射到特征向量中,从而实现对辐射源个体的有效识别,而且该特 征参数对噪声干扰不敏感,在信噪比为20 dB时,系统识别率达到95%以上,在信噪比为5 dB 时系统识别率仍优于80%,验证了所提方法的有效性。     

大规模逻辑神经网络印刷体汉字识别系统

逻辑神经网络是一种采用快速学习算法、RAM阵列实现的数字网络。本文描述了采用这种网络模型实现的印刷体汉字识别系统。这是一个初步实用的系统, 可识别大约4000个不同字号的宋体汉字及其它字符, 其识别率达99%, 甚至对于实际书刊, 其识别率也能达到95%左右。系统使用了大约384,000个神经节点, 是一个复杂的大规模神经网络。和其它同类系统相比, 具有适应性、稳固性好、学习速度快以及可用数字集成电路全硬件并行实现等优点。     

基于计算机视觉技术的鲜蛋等级评定系统

建立了一个基于计算机视觉技术检测鸡蛋的内外品质特性,采用LM神经网络进行分类识别的系统.该系统通过提取鸡蛋的蛋型指数、裂纹及污垢、蛋形尺寸等特征,实现利用神经网络的最优模型完成鸡蛋的分级检测,正确识别率达到87%.     

基于粗糙集及RBF网络的英文字母识别

将粗糙集理论与神经网络相结合,针对7×5分辨率的大写英文字母,构建了基于RBF网络的字母识别系统,给出了该识别系统的核心算法与核心结构.该系统利用粗糙集中最小决策算法对识别矩阵进行属性约简,减少了大量的计算和数据库存储量,同时提高了系统识别速度和识别率.通过计算机模拟实验,将该识别系统的识别率与标准BP网络算法及改进BP网络算法相比较,证实了该系统的优越性,在有约1/7的像素点受到随机干扰的情况下,该系统识别率仍可达到88%以上.     

An evolutionary decoding method for HMM-based continuous speech recognition systems using particle swarm optimization

The main recognition procedure in modern HMM-based continuous speech recognition systems is Viterbi algorithm. Viterbi algorithm finds out the best acoustic sequence according to input speech in the search space using dynamic programming. In this paper, dynamic programming is replaced by a search method which is based on particle swarm optimization. The major idea is focused on generating initial population of particles as the speech segmentation vectors. The particles try to achieve the best segmentation by an updating method during iterations. In this paper, a new method of particles representation and recognition process is introduced which is consistent with the nature of continuous speech recognition. The idea was tested on bi-phone recognition and continuous speech recognition workbenches and the results show that the proposed search method reaches the performance of the Viterbi segmentation algorithm ; however, there is a slight degradation in the accuracy rate.     

The use of speed-up techniques for a speech recognizer system

In speech recognition, not just the accuracy of an automatic speech recognition application is important, but also its speed. However, if we want to create a real-time speech recognizer, this requirement limits the time that is spent on searching for the best hypothesis, which can even affect the recognition accuracy. Thus the applied search method plays an important role in the speech recognition task, and so does its efficiency, i.e. how quickly it finds the uttered words. To speed up this search process, various ideas are available in the literature: we can use search heuristics, multi-pass search, or apply a family of aggregation operators. In this paper we test all these methods in turn, and combine them with a set of other novel speed-up ideas. The test results confirm that all of these techniques are valuable: using combinations of them helped make the speech recognition process over 12 times faster than the basic multi-stack decoding algorithm, and almost 11 times faster than the Viterbi beam search method.     

基于语音识别和手机平台的英语口语发音学习系统

研究一种实际可行的手机平台上基于语音识别技术的英语学习系统的应用方案。系统主要以HMM(隐马尔可夫模型)和Viterbi算法作为模型和算法基础,同时针对手机平台的限制,在算法设计和实现方面进行改进,达到降低运算时间同时保证识别精度的目的。     

Super-human multi-talker speech recognition: A graphical modeling approach

We present a system that can separate and recognize the simultaneous speech of two people recorded in a single channel. Applied to the monaural speech separation and recognition challenge, the system out-performed all other participants – including human listeners – with an overall recognition error rate of 21.6%, compared to the human error rate of 22.3%. The system consists of a speaker recognizer, a model-based speech separation module, and a speech recognizer. For the separation models we explored a range of speech models that incorporate different levels of constraints on temporal dynamics to help infer the source speech signals. The system achieves its best performance when the model of temporal dynamics closely captures the grammatical constraints of the task. For inference, we compare a 2-D Viterbi algorithm and two loopy belief-propagation algorithms. We show how belief-propagation reduces the complexity of temporal inference from exponential to linear in the number of sources and the size of the language model. The best belief-propagation method results in nearly the same recognition error rate as exact inference.     

Phonetic segmentation using multiple speech features

In this paper we propose a method for improving the performance of the segmentation of speech waveforms to phonetic units. The proposed method is based on the well known Viterbi time-alignment algorithm and utilizes the phonetic boundary predictions from multiple speech parameterization techniques. Specifically, we utilize the most appropriate, with respect to boundary type, phone transition position prediction as initial point to start Viterbi time-alignment for the prediction of the successor phonetic boundary. The proposed method was evaluated on the TIMIT database, with the exploitation of several, well known in the area of speech processing, Fourier-based and wavelet-based speech parameterization algorithms. The experimental results for the tolerance of 20 milliseconds indicated an improvement of the absolute segmentation accuracy of approximately 0.70%, when compared to the baseline speech segmentation scheme.     

基于汉语语音特点的大词表语音识别系统的研究

本文探讨了汉语语音识别的若干问题,并简单介绍了一个大词表汉语语音识别系统,该系统充分考虑了汉语语音的特点,其中主要是汉语语音具有音节性比较强的特点、音节的简单声韵母结构以及汉语以词/词组为语音交流基础的特点.该系统一个显著的特点是系统可以不进行任何训练地添加新词汇,从而使得系统具有比较好的用户接口. 现在系统具有10,000多个词汇,实时测试的平均识别结果是93.1％.     

Application of hidden Markov models to automatic speech endpoint detection

Accurate detection of the boundaries of a speech utterance during a recording interval has been shown to be crucial for reliable and robust automatic speech recognition. The endpoint detection problem is fairly straightforward for high-level speech signals spoken in low-level stationary noise environments (e.g. signal-to-noise ratios greater than 30 dB). However, these ideal conditions do not always exist. One example, where reliable word detection is difficult, is speech spoken in a mobile environment. Because of road, tire, fan noises, etc. detection of speech often becomes problematic.Currently, most endpoint detection algorithms use only signal energy and duration information to perform the endpoint detection task. These algorithms perform quite well with reasonable signal-to-noise ratios. However, under the harshest of conditions (e.g. in a car travelling at 60 mph with the fan on high) these algorithms begin to fail.In this paper, an endpoint detection algorithm is presented which is based on hidden Markov model (HMM) technology. The algorithm explicitly determines a set of speech endpoints based on the output of a Viterbi decoding algorithm. This algorithm was tested using a template-based speech recognition system and also using an HMM based system.Based on a speaker dependent speech database from four talkers, recorded in a mobile environment under five different driving conditions (including traveling at 60 mph with the fan on), we tested several endpoint detection schemes. The results showed that, under some conditions, the HMM-based approach to endpoint detection performed significantly better than the energy-based system. The overall accuracy of the system using the HMM endpoint detector, when trained with clean inputs and when tested on the 11 word digits vocabulary (zero through nine and oh) with speech recorded in various mobile environments, was 99.7%. The equivalent accuracy of the energy based endpoint detector was 95.2% in a template based recognizer.     

格式正确的有限命令识别

有些应用场合中语音识别系统的待识别词表确定,但对识别结果要求严格,针对这样的应用,提出了一种语法规则——句序字位规则,并且给出了一种基于连续语音识别的Viterbi算法和句序字位法的语音识别算法。最后针对一组指挥命令进行了实验,实现了格式正确的有限命令识别。     

衡阳方言孤立词识别研究

目前,汉语识别已经取得了一定的研究成果.但由于中国的地域性差异,十里不同音,使得汉语识别系统在进行方言识别时识别率低、性能差.针对语音识别系统在对方言进行识别时的缺陷,构建了基于HTK的衡阳方言孤立词识别系统.该系统使用HTK3.4.1工具箱,以音素为基本识别单元,提取39维梅尔频率倒谱系数（MFCC）语音特征参数,构建隐马尔可夫模型（HMM）,采用Viterbi算法进行模型训练和匹配,实现了衡阳方言孤立词语音识别.通过对比实验,比较了在不同因素模型下和不同高斯混合数下系统的性能.实验结果表明,将39维MFCC和5个高斯混合数与HMM模型结合实验时,系统的性能得到很大的改善.