检索结果-内蒙古大学图书馆

您好，读者！请登录

内蒙古大学图书馆

首页
概况
党建
资源
服务
科研支持
- 论文收录引用证明
- 科技查新
知识产权
档案馆
帮助

咨询与建议

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

您的常用邮箱：*

您的手机号码：*

问题描述：

当前已输入0个字，您还可以输入200个字

全部搜索
期刊论文
图书
学位论文
标准
纸本馆藏
外文资源发现
数据库导航
超星发现

高级检索

时间限定

出版年份：

文献类型

图书期刊文献学位论文多媒体

馆藏选择

电子馆藏纸本馆藏

核心期刊

全部期刊 SCI 收录期刊 SSCI 收录期刊 EI 收录期刊 CSCD 收录期刊 CSSCI 收录期刊

语言

中文英文

文献类型

期刊文献图书学位论文标准纸本馆藏

帮助

文字说明：

T=题名（书名、题名），A=作者（责任者），K=主题词，P=出版物名称，PU=出版社名称，O=机构（作者单位、学位授予单位、专利申请人），L=中图分类号，C=学科分类号，U=全部字段，Y=年（出版发行年、学位年度、标准发布年）

检索规则说明：

AND代表“并且”；OR代表“或者”；NOT代表“不包含”；(注意必须大写,运算符两边需空一格)

检索范例：

范例一：(K=图书馆学 OR K=情报学) AND A=范并思 AND Y=1982-2016
范例二：P=计算机应用与软件 AND (U=C++ OR U=Basic) NOT K=Visual AND Y=2011-2016

分类表

所选分类

>> <<

限定检索结果

文献类型

2,046 篇 会议
407 篇 期刊文献
4 篇 学位论文
1 册 图书

馆藏范围

2,458 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

602 篇 工学
- 443 篇 电气工程
- 131 篇 计算机科学与技术...
- 108 篇 信息与通信工程
- 79 篇 电子科学与技术（可...
- 24 篇 软件工程
- 23 篇 生物医学工程（可授...
- 21 篇 控制科学与工程
- 19 篇 仪器科学与技术
- 7 篇 机械工程
- 7 篇 船舶与海洋工程
- 5 篇 材料科学与工程（可...
- 4 篇 交通运输工程
- 3 篇 力学（可授工学、理...
- 3 篇 测绘科学与技术
- 3 篇 石油与天然气工程
- 3 篇 网络空间安全
- 2 篇 光学工程
235 篇 理学
- 204 篇 物理学
- 16 篇 系统科学
- 14 篇 生物学
- 4 篇 数学
- 3 篇 化学
- 2 篇 地球物理学
- 2 篇 统计学（可授理学、...
52 篇 医学
- 43 篇 临床医学
- 10 篇 特种医学
- 5 篇 基础医学(可授医学...
14 篇 管理学
- 14 篇 管理科学与工程(可...
10 篇 文学
- 5 篇 外国语言文学
- 2 篇 新闻传播学
3 篇 农学
3 篇 军事学
2 篇 教育学
2 篇 艺术学
1 篇 法学

主题

2,458 篇 linear predictiv...
771 篇 speech coding
701 篇 speech analysis
545 篇 speech synthesis
409 篇 bit rate
402 篇 speech recogniti...
347 篇 speech processin...
323 篇 filters
278 篇 frequency
267 篇 testing
252 篇 vocoders
238 篇 vector quantizat...
236 篇 speech enhanceme...
224 篇 quantization
213 篇 predictive model...
201 篇 signal processin...
182 篇 speech
182 篇 cepstral analysi...
179 篇 signal synthesis
177 篇 signal analysis

机构

32 篇 school of electr...
23 篇 at and t bell la...
22 篇 bolt beranek and...
19 篇 bell laboratorie...
11 篇 at and t bell la...
10 篇 naval research l...
10 篇 department of el...
10 篇 ntt human interf...
10 篇 lpcs enserg gren...
9 篇 lpcs-enserg gren...
8 篇 centre for commu...
8 篇 signal technolog...
8 篇 department of el...
8 篇 cselt torino
8 篇 acoustics resear...
7 篇 electrical engin...
7 篇 acoustics resear...
7 篇 comsat laborator...
7 篇 central research...
7 篇 inrs-télécommuni...

作者

19 篇 s. cristoloveanu
18 篇 l. rabiner
14 篇 a.m. kondoz
14 篇 j. makhoul
13 篇 t. barnwell
13 篇 p. mermelstein
12 篇 t. moriya
12 篇 b. atal
12 篇 a. gersho
12 篇 p. hedelin
11 篇 g. ghibaudo
11 篇 gray rm
10 篇 a. rosenberg
10 篇 rabiner lr
10 篇 k. ozawa
9 篇 t.p. barnwell
9 篇 p. kabal
9 篇 a. mccree
9 篇 b.g. evans
8 篇 v. viswanathan

语言

2,284 篇 英文
167 篇 其他
4 篇 中文
3 篇 土耳其文

检索条件"主题词=Linear predictive coding"

共 2458 条记录，以下是461-470 订阅

全选清除本页清除全部题录导出标记到"检索档案"

详细简洁

排序：

Optimizing Vowel Recognition in Thai Spoken Language using Reduced LPC Spectrum and Reduced Feature Set of Critical Band Intensities

Optimizing Vowel Recognition in Thai Spoken Language using R...

引用

International Symposium on Communications and Information Technologies (ISCIT)

作者： Nongnuch Suktangman Kham Khanthavivone Kraisin Songwatana Faculty of Engineering King Mongkut''s Institute of Technology Ladkrabang Bangkok Thailand Thailand

This paper presents an optimization for vowel recognition in Thai spoken language. Thai language consists of 18 unmixed vowels (a,aa,i,ii,omega,omegaomega,u,uu,e,ee,epsi,epsiepsi,o,oo, gamma,gammagamma,sigmav,sigmavsigmav); and 6 mixed vowels (ua,u:a:,ia,i:a:,omegaa,omega:a:). Previously we have proposed use a 3-stage decision-making process: using voice energy and polynomial regression for short and long vowels classification; identifying voice segment (frame) into basic vowels; and using decision rule for recognition on unmixed and mixed vowels. The optimization is done by reducing the order of LPC and choosing a reduced set of critical band intensities (CBI). With this, the percentage accuracy remains high while processing time is decreased by 85%. The experiment is carried out on voice samples from 3 male model, 3 female model, and 2 male and 2 female model, yielding an accuracy of 91.47, 94.98, and 91.72 percents respectively

关键词： Speech recognition Natural languages linear predictive coding Equations Speech analysis Polynomials Transfer functions Humans Power engineering and energy Decision making

来源：评论

学校读者我要写书评

暂无评论

Improvement of Esophageal Speech using LPC and LF Model

Improvement of Esophageal Speech using LPC and LF Model

引用

International Conference on Biomedical and Pharmaceutical Engineering, ICBPE

作者： R. Sirichokswad P. Chanyagorn W. Charoensuk P. Boonpramuk N Kasemkosin H.H. Szu International Biomedical Engineering Programme Mahidol University Thailand Dept. Electrical Engineering Mahidol University Thailand Dept. of Control Systems and Instrumentation Engineering KMUTT Thailand Audiology and Speech Clinic Mahidol University Thailand Office of U.S. Naval Research Virginia USA

Esophageal speech is a restoration of speech communication in laryngectomized patient. Due to irregular pharyngoesophageal (PE) segment vibration and aerodynamic limitation, the esophageal phonation provides higher volatility of the fundamental frequency ( f 0 ) compared with laryngeal phonation. It is difficult to determine an accurate f 0 in esophageal speech. This paper focuses on algorithms for f 0 modification. linear predictive coding (LPC) and autocorrelation function are used to calculate the f 0 They are well performed in the normal case. However, the determination results of f 0 in esophageal speech are highly unstable without any modification of a conventional LPC technique. By proposing a smoothing technique, an accurate f 0 and pitch period in esophageal speech can be determined and used in LF model. Experimental results from 18 subjects suggest that average f 0 of esophageal phonation is lower than laryngeal phonation. The speech synthesized using this proposing technique produced better sound quality than un-processed esophageal speech.

关键词： Esophagus linear predictive coding Autocorrelation Speech synthesis Filters Smoothing methods Speech enhancement Speech analysis

来源：评论

学校读者我要写书评

暂无评论

Dynamic Scaling of Encoded Speech Through the Direct Modification of Coded Parameters

Dynamic Scaling of Encoded Speech Through the Direct Modific...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： R.A. Sukkar R. Younce Peng Zhang Tellabs Inc. Naperville IL USA

In new generation networks like 3G wireless and VoIP, a great deal of emphasis is put on transcoder-free operation (TrFO), where speech remains coded throughout the core network. Any network-based speech processing function must, therefore, operate on the coded parameters directly if the value of TrFO is to be realized. Many of these functions, like echo control, noise reduction, and gain control can be viewed as dynamic amplitude scaling of the speech signal. Given that an intermediate step of decoding/re-encoding is not an option, we present, in this paper, a method for dynamic scaling of speech in the coded-domain directly. We derive expressions for modifying the relevant coded parameters such that the resulting decoded speech would correspond to the desired scaled signal. Experimentally, we use the AMR 12.2 kbps coder, and show that the proposed method results in a signal whose level, as well as speech quality, closely matches the desired scaled signal

关键词： Decoding Noise reduction Speech processing Speech enhancement Gain control Telephone sets Signal processing Performance gain linear predictive coding Degradation

来源：评论

学校读者我要写书评

暂无评论

Analysis and Synthesis of Vowels Using Matlab

Analysis and Synthesis of Vowels Using Matlab

引用

IEEE International Conference on Automation, Quality and Testing, Robotics, AQTR

作者： Alina Nica Alexandru Caruntu Gavril Toderean Ovidiu Buza Department of Telecommunications Technical University of Cluj Napoca Cluj-Napoca Romania

We propose a software environment in Matlab, in order to extract the main features from the Romanian vowels and to synthesize the vowels. The used analysis techniques for the estimation of the parameters are: time domain analysis for energy and zero-crossing rate (ZCR), cepstral analysis for fundamental frequency and linear predictive coding (LPC) for formants. Also, to synthesize the vowels, we used the LPC method

关键词： Speech analysis Speech synthesis Signal analysis Cepstral analysis Signal synthesis linear predictive coding Spectrogram Feature extraction Frequency estimation Time domain analysis

来源：评论

学校读者我要写书评

暂无评论

Speech Stream Detection in Strong Noise based on linear Prediction

Speech Stream Detection in Strong Noise based on Linear Pred...

引用

IEEE Conference on Industrial Electronics and Applications (ICIEA)

作者： Rubo Zhang Tian Wu Xueyao Li Dong Xu College of Computer Science and Technology Harbin Engineering of Technology Harbin Heilongjiang China

The speech signal is usually mixed with a great deal of noise, and the noise weakens seriously the performance of the algorithms to detect speech signal. This paper presents a robust algorithm for speech signal detection in low SNR based on the linear prediction technology. The proposed approach firstly decreases noise by using linear predication residual, then decides whether speech signal is contained or not according to the coefficients statistics of LPC. The operation to speech component is only taken in prediction residual by enhancement, which produces little impairment to the formant of the speech. As most of the energy components of speech signals exist in the region between 300 Hz and 3000 Hz, only those LPC coefficients in this region are taken into account, which also reduces the influence from noise. Experiments show that it is particularly immunized for the proposed algorithm to the strong noise, especially in the white noise. At last, the performance of the algorithm is compared to the approach based on short-term energy in various noise condition, and quantified using the probability of correct classification. The results show that the proposed algorithm has an overall better performance than the referred approach, such as white noise and factory noise to low SNR

关键词： Speech enhancement Signal processing algorithms Signal detection Speech coding Speech processing linear predictive coding Detection algorithms Signal to noise ratio White noise Working environment noise

来源：评论

学校读者我要写书评

暂无评论

Set-membership filtering strategies for multipulse coding

Set-membership filtering strategies for multipulse coding

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： D. Joachim R. Salmon J.R. Deller Electrical Engineering and Computer Science Tulane University USA Electrical and Computer Engineering Michigan State University USA

This paper describes a new class of multi-pulse coders based on bounded-error identification. Multi-pulse coders provide effective representations of speech signals by extracting linear prediction parameters and approximating the effect of excitation signals with strategic sparse impulse sequences. These sparse impulses can be derived according to a set-membership optimization set volume reduction strategy. The proposed method offers the added benefit of improved scalar quantization through the choice of coder parameters from a feasibility set

关键词： Filtering Speech analysis Speech coding Speech processing linear predictive coding Pulse generation predictive models Signal processing Quantization Approximation algorithms

来源：评论

学校读者我要写书评

暂无评论

Adaptive Non-linear Prediction for Speech Signals in Mixture Noise Environments

Adaptive Non-Linear Prediction for Speech Signals in Mixture...

引用

IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： Hirobumi Tanaka Yuichiroh Ohhashi Tetsuya Shimamura Graduate School of Science and Engineering Saitama University Saitama Japan Derparment of Inforrmation and Computer Sciences Saitama University Saitama Japan

For robust prediction analysis for speech signals in impulsive noise environments, we had proposed the OSLMS with AS predictor in Y. Ohhashi et al. (2004). Assuming general noise environments, however, we cannot neglect background noises. Therefore, in this paper, we saddled properties of the LPC in a mixture noise environment which has a content of impulsive noises and white Gaussian noises, and prove that an adaptive predictor perform more effectively in the mixture noises which has white noises with low power than batch predictors. Furthermore, we conducted prediction experiments using 3 adaptive predictors including the OSLMS with AS in the mixture noise environments. As a result, She OSLMS with AS predictor provided good performance in mixture noise environments which has practical white noise power, too

关键词： Speech enhancement Working environment noise Gaussian noise linear predictive coding White noise Speech processing Sorting Signal processing Speech analysis Background noise

来源：评论

学校读者我要写书评

暂无评论

Speaker Identification System Based on Multi-Classifier

Speaker Identification System Based on Multi-Classifier

引用

World Congress on Intelligent Control and Automation (WCICA)

作者： Bo Wang Yiqiong Xu Bicheng Li Department of Information Science Information and Engineering University Zhengzhou China

This paper presents a practical speaker recognition system based on multi-classifier structure. Multi-classifier structure overcomes the shortcomings of single classifier, such as low recognition rate, narrow application field and critical demand of environment. Additionally, multi-classifier provides a novel way of improvement of system performance. The involved classifiers include ANN (artificial neural networks), GMM (Gaussian mixed model), sub-band classifiers, etc. The input features of classifiers contain MFCC (mel frequency cepstrum coefficient), LPCC (linear prediction cepstrum coefficient). Multi-classifier confusion adopts CFM (classification figure of merit) principle as object function. In practical application, the recognition rate of the system achieves 94% in environment of super short wave (SNR 15db)

关键词： linear predictive coding Artificial neural networks Cepstrum Information science Speaker recognition System performance Mel frequency cepstral coefficient Intelligent control Automation

来源：评论

学校读者我要写书评

暂无评论

Influence of the LPC Filter Upon the Perception of Breathiness and Vocal Effort

Influence of the LPC Filter Upon the Perception of Breathine...

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Karl I . Nordstrom Peter F. Driessen Glen A. Rutledge Department of Electrical and Computer Engineering University of Victoria Victoria Canada DB Research Limited Victoria Canada

According to the source-filter paradigm, the perception of breathiness and vocal effort should be primarily controlled by the glottal source and be little affected by the formant filter. This experiment investigates whether the formant filter estimated by linear prediction (LPC) can influence the perception of breathiness and vocal effort. The experiment starts with a pair of voice samples. One sample exhibits high effort and the other sample exhibits breathiness. LPC estimates a filter and residual for each sample. The influence of the residual is eliminated by providing both filters with the same artificial source during resynthesis. The synthesized samples differ only according to the difference between the two filters. Three pairs of samples were evaluated by seven people in listening tests. The results demonstrate that the LPC filters do influence the perception of breathiness and vocal effort. When a voice changes between breathiness and vocal effort, the spectral envelope changes. This change is captured by the LPC filter rather than the residual. A closer look at the LPC algorithm provides an explanation for this result

关键词： linear predictive coding Frequency Shape Pulse shaping methods Power harmonic filters Signal processing Information technology Information filtering Information filters Nonlinear filters

来源：评论

学校读者我要写书评

暂无评论

Frame Change Ratio: A Measure to Model Short-Time Stationarity of Speech

Frame Change Ratio: A Measure to Model Short-Time Stationari...

引用

International Conference on Innovations in Information Technology

作者： M. Sarwar Ehsan Gernot Kubin Signal Processing and Speech Communication Laboratory Graz University of Technology Austria

ISBN: (纸本)1424406730;1424406749

In many applications such as speech coding and PLC (packet loss concealment) methods, short-time stationarity of speech is assumed. In this paper, we introduce a new measure to model the short-time stationarity of speech. The model is developed by using a large amount of speech data together with real speech codecs and PLC methods. Experiments show that a single parameter, the frame change ratio, can be used to model this characteristic. This parameter is independent of speech material, speaker gender, and loss rate. It provides generic characteristics of the deviation of speech from stationary behavior when evolving from one frame to the next

关键词： Programmable control Speech processing Speech coding Speech codecs Rate-distortion linear predictive coding Random processes Loss measurement Signal processing Oral communication

来源：评论

学校读者我要写书评

暂无评论

没有更多数据了...

全选清除本页清除全部题录导出标记到“检索档案”

共246页 << < 43 44 45 46 47 48 49 50 51 52 > >>

检索报告对象比较合并检索0

隐藏清空

合并搜索

回到顶部

执行限定条件

内容：

评分：

请选择保存的检索档案：

请选择收藏分类：

订阅名称：

通借通还

温馨提示：

图书名称：

借书校区：

取书校区：

手机号码：

邮箱地址：

一卡通帐号：

电话和邮箱必须正确填写，我们会与您联系确认。

联系人：

所在院系：

联系邮箱：

联系电话：

内蒙古自治区呼和浩特市赛罕区大学西街235号邮编: 010021

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：