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2311-2320 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：

Word recognition and encoding using poles

Word recognition and encoding using poles

引用

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

作者： D. Tuffelli B. Groc E.N.S. d'Electronique et de radioélectricite Grenoble

At learning, LPC is used to get the reference poles corresponding to the words. During the recognition, the order of the filtering is variable and imposed by the dictionary. The distance between an input speech window and a dictionary speech window is computed with a method near Itakura's method but using a series of two-order inverse filtering. An improved dynamic programming is used allowing parallel computation for several words.

关键词： Encoding Dictionaries Dynamic programming Speech analysis Concurrent computing linear predictive coding predictive coding Speech coding Particle measurements Character recognition

来源：评论

学校读者我要写书评

暂无评论

Improving Lpc Spectral Envelope Extraction Of Voiced Speech By True-Envelope Estimation

Improving Lpc Spectral Envelope Extraction Of Voiced Speech ...

引用

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

作者： F. Villavicencio A. Robel X. Rodet Analysis-Synthesis team I. R. C. A. M. Paris France

n this work we address the problem of all pole spectral envelope estimation for speech signals. The currently widely used all pole spectral envelope model suffers from well-known systematic errors and more severely from model order mismatch. We will propose a procedure to first establish a band limited interpolation of the observed spectrum using a recently rediscovered true envelope estimator and then using the band limited envelope to derive an all pole envelope model named TE-LPC . The band-limited envelope that is used to derive the all pole envelope model reduces the problem of the unknown all pole model order. For the experimental investigation we propose a new perceptually motivated residual spectral peak flatness measure. The experimental results demonstrate that the proposed method significantly increases the spectral flatness for the perceptually especially important low order harmonics of voiced utterances

关键词： linear predictive coding Power harmonic filters Interpolation Speech analysis Signal analysis Data mining Speech coding Production systems Polynomials Frequency

来源：评论

学校读者我要写书评

暂无评论

Frequency weighted linear prediction

Frequency weighted linear prediction

引用

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

作者： P. Chu D. Messerschmitt Department of Electrical Engineering and Conputer Science University of California Berkeley CA USA

A new spectral distance measure is defined by inserting a multiplicative frequency weighting term into the conventional Itakura-Saito measure. Then the weighting function is a simple one pole function, then the minimization of the distance between a signal spectrum and an arbitrary N-pole filter results in a set of linear equations that is symmetric and solvable by Cholesky decomposition. When the weighting function is a multiple pole function, the resulting spectral distance minimization produces a set of nonlinear algebraic equations, but fortunately a simple method exists for obtaining an approximate solution which can be refined using the Newton- Raphson method. Results of some preliminary trials in applying the technique to LPC vocoding are described.

关键词： Frequency measurement linear predictive coding Filters Nonlinear equations Speech Autocorrelation Fourier transforms Weight measurement Frequency estimation Gain measurement

来源：评论

学校读者我要写书评

暂无评论

CCD adaptive filtering for robust LPC speech processing

CCD adaptive filtering for robust LPC speech processing

引用

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

作者： G. Borsuk M. White Westinghouse Advanced Technology Laboratories Baltimore MD USA

A monolithic CCD adaptive filter chip is described which implements the Widrow-Hoff "clipped-data" LMS adaptive algorithm. The chip can be used as a pre-filter noise canceller, analysis filter, or pre-whiten... 详细信息

关键词： Charge coupled devices Adaptive filters linear predictive coding Speech processing Noise robustness Least squares approximation Adaptive algorithm Noise cancellation Speech analysis Nonlinear filters

来源：评论

学校读者我要写书评

暂无评论

An 8 kbit/s ACELP coder with improved background noise performance

An 8 kbit/s ACELP coder with improved background noise perfo...

引用

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

作者： R. Hagen E. Ekudden Research Ericsson Radio Systems Audio and Visual Technology Sweden

This paper describes an 8 kbit/s ACELP speech coder with high performance for both speech and non-speech signals such as background noise. While the traditional waveform matching LPAS structure employed in many existing speech coders provides high quality for speech signals, it has significant performance limitations for, for example, background noise. The coder presented here employs a novel adaptive gain coding technique using energy matching in combination with a traditional waveform matching criterion providing high quality for both speech and background noise. The coder has a basic structure similar to that of the 7.4 kbit/s D-AMPS EFR coder, with a 10 th order LPC, high resolution adaptive codebook and a 4 pulse algebraic codebook. The performance for speech signals is equivalent to or better than that of state-of-the-art 8 kbit/s coders, while for background noise conditions the performance is significantly improved.

关键词： Background noise Speech enhancement Speech synthesis Speech coding Telephony Speech analysis linear predictive coding Energy resolution Signal resolution Mobile communication

来源：评论

学校读者我要写书评

暂无评论

Speaker-independent phonation recognition for Malay plosives using neural networks

Speaker-independent phonation recognition for Malay plosives...

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Hua Nong Ting J. Yunus S.H. Salleh Faculty of Electrical Engineering Universiti Teknologi Malaysia Skudai Johor Malaysia

The paper investigates the use of neural networks in recognizing the phonation of the speech sounds. The proposed method classifies the Malay plosive sounds of adults and children based on phonation in a speaker-indep... 详细信息

关键词： Neural networks Testing Loudspeakers linear predictive coding Hidden Markov models Spatial databases Feature extraction Speech coding Speech recognition Pattern recognition

来源：评论

学校读者我要写书评

暂无评论

Tone modifications used for improving the discrimination of Mandarin words for the hearing-impaired

Tone modifications used for improving the discrimination of ...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Jian Lu N. Uemi Gang Li T. Ifukube Research Institute for Electronic Science Hokkaido University Sapporo Hokkaido Japan Tianjin University TJU China

In this paper, a digital processing method is described for modifying tone contrast that was defined as the difference in frequencies between peaks and valleys of pitch curves in natural utterances. Speech signals with modified tones were presented to hearing-impaired Chinese listeners who were asked to identify four alternative Mandarin words. Employing this method, it was found that modified speech with enhanced tone contrast contributed moderate gains in the percentage correct word identification when compared to unmodified speech, while reducing tone contrast generally reduced the percentage correct identification. These findings therefore offer support to the assertion that a hearing aid with tone modifications is indeed effective for hearing-impaired Chinese.

关键词： Auditory system Frequency Speech enhancement Signal processing Natural languages Speech processing linear predictive coding Vocoders Speech recognition

来源：评论

学校读者我要写书评

暂无评论

Harmonic+noise coding using improved V/UV mixing and efficient spectral quantization

Harmonic+noise coding using improved V/UV mixing and efficie...

引用

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

作者： E.W.M. Yu Cheung-Fat Chan Department of Electronic Engineerin City University of Hong Kong Kowloon Hong Kong China

This paper presents a harmonic+noise speech coder which uses an efficient spectral quantization technique and a novel voiced/unvoiced (V/UV) mixing model. The harmonic magnitudes are coded at 23 bits/frame using the magnitude response of a linear predictive coding (LPC) system. The difference between the harmonic magnitudes and the sampled magnitude response is minimized by the closed-loop approach. The V/UV mixing is modeled by a smooth function which is derived from the speech spectrum envelope based on the flatness measure. The V/UV mixing model allows noise to be added in the harmonic portion of speech spectrum so that buzzyness is reduced. The V/UV mixing information is determined from the spectral parameters available in the decoder, no bits are needed for transmitting the V/UV information. A 1.4 kbps harmonic coder is developed. The speech quality of the coder is comparable to other harmonic coders operating at higher rates.

关键词： Quantization linear predictive coding Power harmonic filters Frequency Decoding Speech enhancement Speech coding Speech analysis Electronic mail Noise reduction

来源：评论

学校读者我要写书评

暂无评论

On nonlinear utilization of intervector dependency in vector quantization

On nonlinear utilization of intervector dependency in vector...

引用

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

作者： M. Skoglund J. Skoglund Department of Signals Sensors and Systems Royal Institute of Technology Stockholm Sweden Department of Information Theory Chalmers University of Technology Goteborg Sweden

This paper presents an approach to speech vector quantization of sources exhibiting intervector dependency. We present the optimal decoder based on a collection of received indices. We also present the optimal encoder for such decoding. The optimal decoder can be implemented as a table look-up decoder, however the size of the decoder codebook grows very fast with the size of the collection of utilized indices. This leads us to introduce a method for storing an approximation to the set of optimal decoder vectors, based on linear mapping of a block code vector quantization. In this approach a heavily reduced set of parameters is employed to represent the codebook. Furthermore, we illustrate that the proposed scheme has an interpretation as nonlinear predictive quantization. Numerical results indicate high gain over memoryless coding and memory quantization based on linear predictive coding. The results also show that the sub-optimal approach performs close to the optimal.

关键词： Vector quantization Decoding Signal processing Encoding Block codes Zinc Sensor systems linear predictive coding Modems Speech

来源：评论

学校读者我要写书评

暂无评论

Objective speech quality measure for cellular phone

Objective speech quality measure for cellular phone

引用

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

作者： K.H. Lam O.C. Au C.C. Chan K.F. Hui S.F. Lau Department of Electrical and Electronic Engineering Hong Kong University of Science and Technology Kowloon Hong Kong China

Cellular phone network speech quality monitoring is a regular task performed by the cellular service providers. Objective speech quality measures are needed in such tasks to provide a reasonably accurate estimate of subjective quality of the network. We performed an experiment to collect real distorted data, conducted a survey to obtain subjective quality measure of the collected speech samples and studied the statistical correlation of 32 objective speech quality measures with the subjective measures. Four of the objective measures were found to be good. Synchronization was found to be important.

关键词： Speech Cellular phones Distortion measurement Cellular networks Time measurement Auditory system Mean square error methods linear predictive coding Frequency measurement Psychoacoustic models

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 << < 228 229 230 231 232 233 234 235 236 237 > >>

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：