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
427 篇 期刊文献
1 册 图书

馆藏范围

2,474 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

622 篇 工学
- 468 篇 电气工程
- 136 篇 计算机科学与技术...
- 113 篇 信息与通信工程
- 82 篇 电子科学与技术（可...
- 25 篇 生物医学工程（可授...
- 24 篇 软件工程
- 22 篇 控制科学与工程
- 19 篇 仪器科学与技术
- 6 篇 机械工程
- 5 篇 材料科学与工程（可...
- 4 篇 测绘科学与技术
- 3 篇 石油与天然气工程
- 3 篇 船舶与海洋工程
- 3 篇 网络空间安全
- 2 篇 力学（可授工学、理...
- 2 篇 光学工程
- 2 篇 交通运输工程
241 篇 理学
- 208 篇 物理学
- 16 篇 系统科学
- 14 篇 生物学
- 5 篇 数学
- 3 篇 化学
- 3 篇 地球物理学
- 2 篇 统计学（可授理学、...
51 篇 医学
- 42 篇 临床医学
- 11 篇 特种医学
- 4 篇 基础医学(可授医学...
14 篇 管理学
- 14 篇 管理科学与工程(可...
9 篇 文学
- 4 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 艺术学
1 篇 教育学

主题

2,474 篇 linear predictiv...
779 篇 speech coding
706 篇 speech analysis
550 篇 speech synthesis
412 篇 bit rate
410 篇 speech recogniti...
350 篇 speech processin...
329 篇 filters
281 篇 frequency
270 篇 testing
254 篇 vocoders
241 篇 vector quantizat...
238 篇 speech enhanceme...
227 篇 quantization
215 篇 predictive model...
202 篇 signal processin...
184 篇 cepstral analysi...
183 篇 speech
179 篇 signal synthesis
178 篇 signal analysis

机构

32 篇 school of electr...
24 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

2,290 篇 英文
176 篇 其他
4 篇 中文
3 篇 土耳其文

检索条件"主题词=Linear Predictive Coding"

共 2474 条记录，以下是711-720 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：

iLBC - a linear predictive coder with robustness to packet losses

iLBC - a linear predictive coder with robustness to packet l...

引用

IEEE Workshop on Speech coding Proceedings

作者： S.V. Andersen W.B. Kleijn R. Hagen J. Linden M.N. Murthi J. Skoglund Department of Communication Technology Adborg University Denmark Global IP Sound San Francisco USA

In this paper, we discuss the internet low bit rate codec (iLBC) with an emphasis on the frame-independent long-term prediction. The frame-independent long-term prediction is a method to exploit pitch-lag correlations in the encoding of speech without suffering multiple-frame speech degradation in connection with transmission loss. We present mean opinion scores for the iLBC codec and show by means of signal examples how the nature of degradation in a predictive codec based on frame-independent long-term prediction differs from that of traditional CELP codecs.

关键词： Robustness Codecs Bit rate linear predictive coding predictive coding Decoding Internet Speech coding Acoustic noise Degradation

来源：评论

学校读者我要写书评

暂无评论

Phoneme classification using naive Bayes classifier in reconstructed phase space

Phoneme classification using naive Bayes classifier in recon...

引用

IEEE Digital Signal Processing Workshop

作者： Jinjin Ye R.J. Povinelli M.T. Johnson Department of Electrical and Computer Engineering Marquette University Milwaukee WI USA Department of Electrical and Computer Engineering Marquette University Milwaukee WI

ISBN: (纸本)0780381165

A novel method for classifying speech phonemes is presented. Unlike traditional cepstral based methods, this approach uses histograms of reconstructed phase spaces. A naive Bayes classifier uses the probability mass estimates for classification. The approach is verified using isolated fricative, vowel, and nasal phonemes from the TIMIT corpus. The results show that a reconstructed phase space approach is a viable method for classification of phonemes, with the potential for use in a continuous speech recognition system.

关键词： Speech recognition Hidden Markov models Cepstral analysis Heart Delay effects Histograms Speech analysis linear predictive coding Neural networks Spectral analysis

来源：评论

学校读者我要写书评

暂无评论

Implementation of speech recognition algorithm for a 32-bit CPU-based portable device

Implementation of speech recognition algorithm for a 32-bit ...

引用

IEEE International Conference on Consumer Electronics (ICCE)

作者： Suhong Ryu Younim Lee Wonyong Sung School of Electrical Engineering Seoul National University Seoul South Korea

ISBN: (纸本)0780373006

We have implemented a connected word speech recognition algorithm for a handheld educational device, SpeakingPartner. This system is based on a low-cost 32-bit CPU, and equips a graphic LCD for animation, 8 Mbyte of DRAM and 32 MB of NAND flash memory. Several previously known software optimization algorithms, such as fixed-point conversion, loop fusion, loop unrolling, and circular addressing, as well as efficient log-likelihood computation and partial backtracking methods are applied. The implementation results show that the peak computation requirement for a real-time implementation is reduced to about 8 MIPS, which translates that a 60 MHz 32-bit CPU based system can perform the speech recognition while playing animation in a multi-tasking mode.

关键词： Speech recognition Signal processing algorithms Animation Cepstrum Delay Floating-point arithmetic Random access memory Real time systems linear predictive coding Feature extraction

来源：评论

学校读者我要写书评

暂无评论

Design and development of speech-control robotic manipulator arm

Design and development of speech-control robotic manipulator...

引用

International Conference on Control, Automation, Robotics and Vision (ICARCV)

作者： S.-H. Salleh Hong Kai Sze Tan Tian Swee Digital Signal Processing Group Faculty of Electrical Engineering Universiti Teknologi Malaysia Malaysia

This paper presents a speech-control robotic manipulator arm via personal computer and self-built robotic arm. It describes the software we developed to collect, process and train the speech database. First, the specifications of the robotic arm will be explained. The algorithms involved will also be briefly discussed. Finally, the result will show the ability of this system to control robotic manipulator arm with human's voice.

关键词： Manipulators Speech recognition Robot control Service robots linear predictive coding Microcomputers Speech processing Humans Industrial control Virtual manufacturing

来源：评论

学校读者我要写书评

暂无评论

A new developed speech recognition education software in teaching LPC and LPC-derived cepstrum

A new developed speech recognition education software in tea...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Hong Kai Sze S.-H. Salleh A.Z. Sha'ameri Digital Signal Processing Group Faculty of Electrical Engineering Universiti Teknologi Malaysia Malaysia

Speech processing systems are highly complex and teaching students in this subject matter with the underlying technologies can be a challenging task. Thus education software is needed to provide the students with more convenient way to study, perform experiments and develop his/her first speech recognition application without an instructor. The Speech Recognition Virtual Classroom (SRVC) was developed as education software in teaching speech recognition. There are various modules, like signal analysis, dynamic time warping, vector quantization and hidden Markov models (HMM). These modules are integrated into a single environment that allows easy learning through experiments with real speech data.

关键词： Speech recognition Education linear predictive coding Cepstrum Hidden Markov models Speech processing Software performance Application software Signal analysis Vector quantization

来源：评论

学校读者我要写书评

暂无评论

A text-independent speaker identification system using PARCOR and AR model

A text-independent speaker identification system using PARCO...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Chia-Hsiung Liu O.T.-C. Chen Signal and Media Laboratories Department of Electrical Engineering National Chung Cheng University Chiayi Taiwan

In this work, we propose the partial-correlation (PARCOR) coefficients scheme to model the cross areas of the several cylinders from the vocal tract. By using the relationship of the acoustic impedance proportional to the reciprocal of cross areas, the ratios of cross areas between each neighboring cylinders are used to model a speaker's vocal tract. The autoregressive model (AR model) is performed on the speech residual signals, that are produced from the inverse vocal tract transform based on the PARCOR, to generate features. These features with the conventional features from the Mel-Frequency Cepstral Coefficient (MFCC) are used for the identification engine of the Gaussian Mixture Model (GMM). According to our computer analyses in the TIMIT speech database, the proposed system can yield better identification performance than the conventional approach.

关键词： Speech analysis Timing Engine cylinders Cepstral analysis Signal analysis Laboratories Impedance Signal generators linear predictive coding Speech recognition

来源：评论

学校读者我要写书评

暂无评论

Dither-like data hiding in multistage vector quantization of MELP and G.729 speech coding

Dither-like data hiding in multistage vector quantization of...

引用

Asilomar Conference on Signals, Systems & Computers

作者： Pao-Chi Chang Hsin-Min Yu Department of Electrical Eng. National Central University Chungli Taiwan

ISBN: (纸本)0780375769

We present a speech data hiding technique that utilizes the characteristics of multistage vector quantization (MSVQ) and subtractive dithering to maintain high speech reconstruction quality. The last stage of MSVQ is used to store the data to be embedded. Similar to subtractive dithering, the noise-like hidden data is subtracted from the signal at the first stage of the encoder and added back at the MSVQ decoder. As a result, the degradation caused by hiding secret data is significantly reduced compared to the traditional simple substitution method.

关键词： Data encapsulation Vector quantization Speech coding Decoding Speech enhancement Degradation linear predictive coding Art Cryptography Protection

来源：评论

学校读者我要写书评

暂无评论

Split VQ and predictive split VQ of LSP parameters in packet networks

Split VQ and predictive split VQ of LSP parameters in packet...

引用

IEEE Digital Signal Processing Workshop

作者： F. Merazka D. Berkani Ecole Nationale Polytechnique (ENP) Algiers Algeria

ISBN: (纸本)0780381165

In many speech coding systems, the LPC coefficients are transformed to the Line Spectrum Pairs (LSP) parameters which are very effective representation for quantization of the LPC information. LSP representation consumes a large part of the total bit rate of the coder. Typically, the LSP are highly correlated from one frame to the next one, and a considerable reduction in bit rate can be achieved by exploiting this interframe correlation. However, interframe LSP coding can cause error propagation when frame erasures occur. In this paper, we compare the erasure performance of a predictive split vector quantizer to that of split vector quantizer based on the ITU G723.1 standard coder. Our results show that a 25 bits/frame split vector quantizer improves the average of spectral distortion of the 24 bits/frame predictive split vector quantizer based on The ITU G.723.1 for different loss rates.

关键词： Intelligent networks Vector quantization Speech coding linear predictive coding Bit rate Code standards Filters Robustness Source coding predictive models

来源：评论

学校读者我要写书评

暂无评论

Multilayered and columnar competitive networks for spoken word recognition

Multilayered and columnar competitive networks for spoken wo...

引用

International Conference on Neural Information Processing

作者： S. Kurogi T. Sasaki M. Itakura T. Nishida Kyushu Institute of Technology Kitakyushu Japan

We have presented a multilayered and columnar competitive network involving competitive associative nets (CANs) and adaptive vector quantization nets (AVQNs) for spoken word recognition. Although the network has shown good performance in recognition rate, it requires a relatively large calculation time owing to the CANs. So, here, we present a new network replacing the CANs by a conventional feature extractor and an additional AVQNs, where as the feature extractor we use one of the three conventional methods: RLS (recursive least squares) for extracting LPCs, LSL (least squares lattice) for PARCOR coefficients, and normalized LSL for normalized PARCOR coefficients. As a result of experiments, the normalized LSL shows almost the same performance as the original network in recognition rate while reducing the calculation time.

关键词： Feature extraction Least squares methods linear predictive coding Data mining Vector quantization Signal processing Lattices Ear Function approximation Speech processing

来源：评论

学校读者我要写书评

暂无评论

Constant delay and rate coding of speech spectral envelope at 11 bits/frame

Constant delay and rate coding of speech spectral envelope a...

引用

IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： E. Yakhnich Y. Bistritz Department of Electrical Engineering Tel-Aviv University Tel-Aviv Israel

Parametric vocoders are used to attain very low rate compression. A new method for segment coding of a speech spectral envelope is proposed, based on coding of constant length segments. It is a computationally efficient method for LPC (linear predictive coding) parameter coding based on TD (temporal decomposition). It attains average LSD (log-spectrum distortion) less than 1 dB at 11 bits/frame. Thus, it saves 56% of the LPC parameter bit budget which is 26% of the total NELP2400 vocoder bit budget. The method offers constant delay and relatively low complexity.

关键词： Delay Speech coding Quantization linear predictive coding Vectors Vocoders Bit rate Interpolation predictive models Frequency

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248页 << < 68 69 70 71 72 73 74 75 76 77 > >>

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：