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381-390 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：

Development of a 6.6 kV - 1 MVA Transformerless Loop Balance Controller

Development of a 6.6 kV - 1 MVA Transformerless Loop Balance...

引用

Annual IEEE Conference on Power Electronics Specialists (PESC)

作者： Naotaka Okada Masahiro Takasaki Hiromitsu Sakai Shuji Katoh Central Research Institute of Electric Power Industry Komae Tokyo Japan Hitachi and Limited Hitachi Ibaraki Japan

To use distribution networks effectively and ensure the smooth introduction of distributed generations, a loop or mesh structure for distribution systems has been proposed using back to back type loop power flow controller (LPC) by CRIEPI. The installation space of looping equipment is restricted in power distribution systems. It is necessary to miniaturize the equipment. We developed 6.6 kV-1MVA transformerless loop balance controller (LBC) as one approach to miniaturization of LPC. A development of LBC has been entrusted by a national project: "Demonstrative Project on Power Network Technology" from the New Energy and Industrial Technology Development Organization (NEDO).

关键词： Control systems linear predictive coding Substations Electrical equipment industry Load flow Voltage control System testing Centralized control Distributed control Space technology

来源：评论

学校读者我要写书评

暂无评论

linear Prediction with Adaptive Filters by using Clonal Selection Algorithm

Linear Prediction with Adaptive Filters by using Clonal Sele...

引用

IEEE Signal Processing and Communications Applications (SIU)

作者： Tuba Caglikantar Rifat Kurban Jeodezi ve Fotogrametri Mühendisliği Bölümü Erciyes Üniversitesi Kayseri Turkey

In this paper, linear prediction of signals is realized with an adaptive filter structure using a clonal selection algorithm (CLONALG). The performances of gradient based least mean squares method and the clonal selection algorithm are compared. Furthermore, the effect of the adaptive filter order for proposed approach on the performance is examined and with higher order filters smaller prediction errors are obtained. Also, it is seen that the dependence of the clonal selection algorithm to the design parameters is less than the conventional methods.

关键词： Adaptive filters Finite impulse response filter Least mean squares methods Algorithm design and analysis linear predictive coding Least squares approximation Nanoelectromechanical systems Resonance light scattering

来源：评论

学校读者我要写书评

暂无评论

Selected applications of LPLE in speech processing

Selected applications of LPLE in speech processing

引用

IEEE Southeastcon

作者： Norelli Schettini Ravi Sankar Department of Electrical Engineering University of South Florida Tampa FL USA

linear prediction with low-frequency emphasis (LPLE), an all-pole modeling technique which emphasizes the lower frequency range of the input signal, was described by Alku and Backstrom (Alku, 2004). The method is based on first interpreting conventional linear predictive (LP) analyses of successive prediction orders with parallel structures using the concept of symmetric linear prediction. The experiments presented in this work are aimed to show that the LPLE method is well-suited for those speech processing and enhancement applications, where low-order all-pole models with improved modeling of the lowest formants are needed. This is done by replacing the LP block with a LPLE block. Distortion measures using Itakura distance and noise cancellation using a Wiener "type" filter were explored, utilizing the LPLE coefficients instead of LP coefficients.

关键词： Speech processing predictive models Frequency Speech analysis linear predictive coding Distortion measurement Humans Auditory system Nonlinear filters Signal processing

来源：评论

学校读者我要写书评

暂无评论

Applying Tunstall coding in the existing SEED format for Seismographic Data

Applying Tunstall Coding in the existing SEED format for Sei...

引用

Data Compression Conference (DCC)

作者： Edwin S. Hong Shu-Fang Newman Computing and Sof tware Systems Institute of Technology University of Washington Tacoma USA

The Standard for the Exchange of Earthquake Data (SEED) is a commonly used file format for recording seismographic data. The data description language (DDL) is part of the specification of the SEED format; it is used to describe the way in which the data has been encoded in the SEED file. All SEED files contain a piece of DDL code that describes how to read the data contained in the file. In this project, we explore the best possible way of compressing seismographic data losslessly under the constraint that it must be describable in DDL. Since DDL is not a Turing-complete language, it is impossible to implement many standard compression algorithms with it. However, we show it is possible to implement a modified Tunstall code under DDL that produces files that are on average 14% smaller than the traditionally used Steim2 compression technique. We also compare our results with a standard linear predictive coding scheme (which is not implementable in DDL) and show that our technique is on average 18% worse on the same set of files.

关键词： Encoding Statistics Software systems Software standards Earthquakes Compression algorithms linear predictive coding Testing Decoding Data compression

来源：评论

学校读者我要写书评

暂无评论

Multi-agent Based Arabic Speech Recognition

Multi-agent Based Arabic Speech Recognition

引用

Web Intelligence and Intelligent Agent Technology Workshops (WI-IAT)

作者： Muhammad Taha Tarek Helmy Reda Abo Alez Faculty of Science Mathematical Department Cairo University Cairo Egypt College of Computer Science and Engineering King Fahd University of Petroleum and Minerals Dhahran Saudi Arabia Faculty of Engineering Computers & Systems Engneering Department Al-Azhar University Cairo Egypt

This paper presents a novel agent-based design for Arabic speech recognition. We define the Arabic speech recognition as a multi-agent-system where each agent has a specific goal and deals with that goal only. Once all the small tasks are accomplished the big task is too. A number of agents are required in order to recast Arabic speech recognition, namely the feature extraction agent and the pattern classification agent. These agents are detailed in this paper.

关键词： Speech recognition Feature extraction Natural languages Hidden Markov models Automatic speech recognition Intelligent agent Pattern classification linear predictive coding Intelligent sensors Standards development

来源：评论

学校读者我要写书评

暂无评论

Sequential Split Vector Quantization of LSF Parameters using Conditional Pdf

Sequential Split Vector Quantization of LSF Parameters using...

引用

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

作者： Saikat Chatterjee T.V. Sreenivas Department of Electrical Communication Engineering Indian Institute of Science Bangalore India

A better performing product code vector quantization (VQ) method is proposed for coding the line spectrum frequency (LSF) parameters; the method is referred to as sequential split vector quantization (SeSVQ). The split sub-vectors of the full LSF vector are quantized in sequence and thus uses conditional distribution derived from the previous quantized sub-vectors. Unlike the traditional split vector quantization (SVQ) method, SeSVQ exploits the inter sub-vector correlation and thus provides improved rate-distortion performance, but at the expense of higher memory. We investigate the quantization performance of SeSVQ over traditional SVQ and transform domain split VQ (TrSVQ) methods. Compared to SVQ, SeSVQ saves 1 bit and nearly 3 bits, for telephone-band and wide-band speech coding applications respectively.

关键词： Vector quantization linear predictive coding Image coding Wideband Speech analysis Power filters Product codes Frequency Rate-distortion Speech coding

来源：评论

学校读者我要写书评

暂无评论

Recognition of Phonemes In a Continuous Speech Stream By Means of PARCOR Parameter In LPC Vocoder

Recognition of Phonemes In a Continuous Speech Stream By Mea...

引用

Canadian Conference on Electrical and Computer Engineering (CCECE)

作者： Ying Cui Kunio Takaya Department of Electrical & Computer Engineering University of Saskatchewan Saskatoon SAS Canada

linear predictive coding (LPC) has been used to compress and encode speech signals for digital transmission at a low bit rate. LPC determines a FIR system that predicts a speech sample from the past samples by minimizing the squared error between the actual occurrence and the estimated. The coefficients of the FIR system are encoded and sent. At the receiving end, the inverse system called AR model is excited by a random signal to reproduce the encoded speech. The use of LPC can be extended to speech recognition since the FIR coefficients are the condensed information of a speech signal of typically 10ms -30ms. PARCOR parameter associated with LPC that represents a vocal tract model based on a lattice filter structure is considered for speech recognition. The use of FIR coefficients and the frequency response of AR model were previously investigated. [1] This paper reports the method to detect a limited number of phonemes from a continuous stream of speech. A system being developed slides a time window of 16 ms and calculates the PARCOR parameters continuously, feeding them to a classifier. A classifier is a supervised classifier that requires training. The classifier uses the Maximum Likelihood Decision Rule. The training uses TIMIT speech database, which contains the recordings of 630 speakers of 8 major dialects of American English. The classification results of some typical vowel and consonant phonemes segmented from the continuous speech are listed. The vowel and consonant correct classification rate are 65.22% and 93.51%. Overall, They indicate that the PARCOR parameters have the potential capability to characterize the phonemes.

关键词： Speech recognition linear predictive coding Vocoders Finite impulse response filter Speech coding Bit rate Lattices Frequency response Maximum likelihood detection Maximum likelihood estimation

来源：评论

学校读者我要写书评

暂无评论

Pitch modification based on syllable units for voice morphing system

Pitch modification based on syllable units for voice morphin...

引用

IFIP International Conference on Network and Parallel Computing Workshops (NPC Workshops)

作者： Yinqiu Gao Zhen Yang Institute of Signal and Information Processing Nanjing University of Posts and Telecommunications Nanjing China

An innovative scheme of voice morphing is proposed to make the speech of a source speaker sound like uttered by a target speaker. The morphing technique can hide people 's identity, age, gender while chatting and doing other things related to the transformation of speech streams online, which can ensure the privacy on the prevalent internet. Speaker individuality transformation is achieved by altering the spectral envelope and estimating the excitation signal by modifying the fundamental pitch frequency in syllable units of the residual signal of the source speech based on linear prediction coding (LPC) model. The main advantage of this scheme relies in the aspect of having considered the dynamic characteristic of the pitch frequency, not just focusing on the average level, which enhances the performance of the whole conversion system compared with general concepts such as discrete pitch frequency mapping and so on. Moreover, in the aspect of the alignment of line spectral frequencies (LSFs) vectors, an advanced technique based on isolated syllables rather than the general dynamic time warping algorithm (DTW) is introduced. The experimental results show that the system is capable of effectively transforming speaker identity whilst the converted speech maintains high quality.

关键词： Loudspeakers linear predictive coding Frequency estimation Frequency conversion Speech synthesis Parallel processing Signal processing Information processing Telecommunications Training data

来源：评论

学校读者我要写书评

暂无评论

Flexible Quantization of Audio and Speech based on the Autoregressive Model

Flexible Quantization of Audio and Speech based on the Autor...

引用

Asilomar Conference on Signals, Systems & Computers

作者： Alexey Ozerov W. Bastiaan Kleijn ACCESS Linnaeus Center Electrical Engineering KTH Royal Institute of Technology Stockholm Sweden

We describe a coding scheme based on audio and speech quantization with an adaptive quantizer derived from the autoregressive model under high-rate assumptions. The main advantage of this scheme compared to state-of-the-art training-based coders is its flexibility. The scheme can adapt in real time to any particular rate and has a computational complexity independent of the rate. Experiments indicate that, compared with a non-scalable conventional fixed-rate code-excited linear predictive (CELP) coding scheme, our real time scalable coder with scalar quantization performs at least as well in the constrained entropy case, and has nearly identical performance for the constrained resolution case.

关键词： Quantization Shape Speech coding Computational complexity Entropy linear predictive coding Frequency estimation Error analysis Real time systems Statistics

来源：评论

学校读者我要写书评

暂无评论

The application of information fusion in the real-time monitoring system

The application of information fusion in the real-time monit...

引用

International Conference on Information Fusion

作者： Wang Chen Miao Zhenjiang Meng Xiao Institute of Information Science Beijing Jiaotong University Beijing China School of Electronic and Information Engineering Beijing Jiaotong University Beijing China

ISBN: (纸本)9780662478300;0662478304

This paper studies the feasibility of information analysis processing technology, which fuses speech and image together in the real-time monitoring system. It emphasizes particularly on speech analysis and fuses these two technologies in terms of scoring strategy. It also makes some improvement on MFCC feature extraction and proposes a quick MFCC algorithm. The proposed algorithm can reach the requirement of real-time system in case of the high precision. To prove it, this paper compares its algorithm with LPC and FFT. The experiment indicates that the EER of LPC is 13.9% and the EER of FFT is 11.1%, but by using the Quick MFCC the EER is only 4.2%. And compared with the traditional MFCC algorithm, the quick MFCC algorithm reduces the run time greatly while maintaining recognition accuracy of the system. Finally the rate of fusion recognition is about 97.8%, which is a good result for the real-time monitoring system.

关键词： Real time systems Monitoring Mel frequency cepstral coefficient Speech analysis Information analysis Feature extraction Speaker recognition Fuses Speech processing linear predictive coding

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 << < 35 36 37 38 39 40 41 42 43 44 > >>

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：