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
404 篇 期刊文献
4 篇 学位论文
1 册 图书

馆藏范围

2,455 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

600 篇 工学
- 441 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 船舶与海洋工程
- 6 篇 机械工程
- 4 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
236 篇 理学
- 205 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

主题

2,455 篇 linear predictiv...
769 篇 speech coding
702 篇 speech analysis
545 篇 speech synthesis
409 篇 bit rate
403 篇 speech recogniti...
346 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
212 篇 predictive model...
200 篇 signal processin...
183 篇 cepstral analysi...
182 篇 speech
179 篇 signal synthesis
177 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

2,281 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 2455 条记录，以下是2271-2280 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：

Reconstruction of missing speech frames using sub-band excitation

Reconstruction of missing speech frames using sub-band excit...

引用

IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： K. Cluver P. Noll Institut füur Frernmeldetechnik Technische Universität Berlin Berlin Germany

A new reconstruction method for frame erasures in speech transmission is presented which is based on parameterization of the speech signal by means of linear prediction (LPC) and voicing analysis. The problem of generating partially voiced substitute speech signals is solved by performing separate voicing decisions in sub-bands. The method yields considerable improvements compared with silence substitution for frame erasure ratios of up to 10% or even 20%. The combination of the reconstruction method with adaptive speech coders showed virtually the same good results for forward adaptation, whereas a higher degradation is caused by backward-adaptive coders.

关键词： Speech analysis linear predictive coding Reconstruction algorithms Signal generators Signal analysis Personal communication networks Extrapolation Phase change materials Degradation Communication networks

来源：评论

学校读者我要写书评

暂无评论

Efficient coding of LPC parameters using adaptive prefiltering and MSVQ with partially adaptive codebook

Efficient coding of LPC parameters using adaptive prefilteri...

引用

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

作者： Y. Tanaka T. Taniguchi Fujitsu Laboratories Limited Kawasaki Japan

The authors propose an efficient vector quantization scheme and a novel linear predictive coding (LPC) analysis scheme, both of which exploit interframe correlation in the successive spectrum envelope of speech signals. The first quantization scheme proposed is a multistage vector quantization of line spectrum pair (LSP) parameters with a partially adaptive codebook (MSVQ-AC). The second algorithm is an LPC analysis scheme, with closed-loop adaptive prefiltering (LPC-PF), which realized temporary higher-order analysis than the standard LPC with a few additional transmission bits. A combined system of the LPC-PF and two-split, two-stage VQ with the adaptive codebook can quantize tenth-order LSP parameters at around 23 bits/frame, realizing sufficient quality and reasonable complexity.< >

关键词： linear predictive coding Adaptive coding Vector quantization Encoding Delay Speech analysis Bit rate Speech coding Discrete cosine transforms Laboratories

来源：评论

学校读者我要写书评

暂无评论

A real time spectral subtraction based speech enhancement scheme

A real time spectral subtraction based speech enhancement sc...

引用

Canadian Conference on Electrical and Computer Engineering (CCECE)

作者： D. Flogeras R. Doraiswami M.E. Kaye Department of Electrical and Computer En ineering University of New Brunswick Fredericton NB Canada

Various speech enhancement schemes are analyzed in terms of the conflicting real time requirements of computational delay, robustness and accuracy. A spectral subtraction scheme is found to be implementable in real time using the available digital processing board. The tasks such as the computation of the spectral contents of the speech plus noise and the noise, smoothing the estimates, and speech enhancement filtering are implemented in the frequency domain using FFT in view of the computational speed and robustness. A variant of a spectral subtraction scheme is implemented in real time on a DSP board and its performance is evaluated.

关键词： Speech enhancement predictive models Amplitude estimation Noise robustness Filtering Real time systems Frequency estimation linear predictive coding Niobium Speech analysis

来源：评论

学校读者我要写书评

暂无评论

Multi stage matrix quantization for very low bit rate speech coding

Multi stage matrix quantization for very low bit rate speech...

引用

IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC)

作者： S. Ozaydin B. Baykal Department of Electrical & Electronics Engineering Gazi University Ankara Turkey Department of Electrical & Electronics Engineering Middle East Technical University Ankara Turkey

ISBN: (纸本)0780367200

Efficient quantization of linear predictive coding (LPC) filter coefficients play an essential role in very-low-bit-rate speech coding systems. This paper examines a new suboptimal matrix quantization scheme for LPC parameters, called multi-stage matrix quantization (MSMQ), which operates at bit rates between 400 and 800 bit/s. With the new matrix quantization method, using a 22.5 ms LPC analysis frame, spectral distortion about 1 dB is achieved at 800 bit/s. In the proposed coder, line spectral frequency (LSF) parameters of multiple consecutive frames are grouped into a superframe and jointly quantized. The new residual LSF vector quantization scheme gives a bit rate reduction in MSMQ without any additional complexity or storage. The new MSMQ leads into several schemes of various computational complexity/storage characteristics.

关键词： Bit rate Speech coding linear predictive coding Nonlinear filters Vector quantization Spectral analysis Frequency Large scale integration Computational complexity Wireless communication

来源：评论

学校读者我要写书评

暂无评论

A novel LPC synthesis model using a binary pulse source excitation

A novel LPC synthesis model using a binary pulse source exci...

引用

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

作者： D. Lin Bell Communications Research Inc. Morristown NJ USA

It is difficult to produce natural-sounding speech using LPC voice coding techniques (vocoder). The main problem is the simplistic model of the excitation source in terms of the pitch pulses and white noise used in the synthesis. The LPC vocoder requires robust and accurate pitch determination, which is a problem that still has not been completely solved. Recently, a multi-pulse approach for modeling the excitation source has been proposed by Atal. However, this approach requires a substantial increase in the bit rate over conventional LPC. We describe in this paper a novel pulse source model for improving the quality of LPC speech. The binary pulse source excitation is obtained from the speech signal using a nonlinear filtering procedure. Our approach is able to maintain good speech quality even at bit rates of 4.8 kbps or lower.

关键词： linear predictive coding Speech synthesis Bit rate Speech coding Speech analysis Vocoders Signal synthesis Noise robustness Filtering Filters

来源：评论

学校读者我要写书评

暂无评论

Distributed speech recognition with codec parameters

Distributed speech recognition with codec parameters

引用

IEEE Workshop on Automatic Speech Recognition and Understanding

作者： B. Raj J. Migdal R. Singh Mitsubishi Electric Research Laboratories Inc. Cambridge MA USA Massachusetts Institute of Technology Cambridge MA USA Carnegie Mellon University Pittsburgh PA USA

Communication devices which perform distributed speech recognition (DSR) tasks currently transmit standardized coded parameters of speech signals. Recognition features are extracted from signals reconstructed using these on a remote server. Since reconstruction losses degrade recognition performance, proposals are being considered to standardize DSR-codecs which derive recognition features, to be transmitted and used directly for recognition. However, such a codec must be embedded on the transmitting device, along with its current standard codec. Performing recognition using codec bitstreams avoids these complications: no additional feature-extraction mechanism is required on the device, and there are no reconstruction losses on the server. We propose an LDA-based method for extracting optimal feature sets from codec bitstreams and demonstrate that features so derived result in improved recognition performance for the LPC, GSM and CELP codecs. For GSM and CELP, we show that the performance is comparable to that with uncoded speech and standard DSR-codec features.

关键词： Speech recognition Codecs Feature extraction GSM Propagation losses Degradation Performance loss Proposals Code standards linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

HMM/NN hybrids for continuous speech recognition

HMM/NN hybrids for continuous speech recognition

引用

National Conference on Radio Science (NSRC)

作者： O.A. Abdel Alim N. Elboghdadly N.M. El Shaar Faculty of Engineering EE . Department Aexaadria University Egypt

ISBN: (纸本)9775031680

The main goal of automatic speech recognition (ASR) is to produce a machine which will recognize accurately normal human speech from any speaker. The recognition system may be classified as speaker-dependent or speaker-independent and isolated-word or connected word. There are three approaches to research in automatic speech recognition (ASR); the acoustic-phonetic approach, the pattern recognition approach, and the database statistical approach. Two approaches of this kind: hidden Markov model (HMM) and artificial neural network (ANN) are presented in this paper.

关键词： Hidden Markov models Neural networks Speech recognition Automatic speech recognition Pattern recognition linear predictive coding Artificial neural networks Speech analysis Spectral analysis Loudspeakers

来源：评论

学校读者我要写书评

暂无评论

Quality comparison measure for linear predictive systems

Quality comparison measure for linear predictive systems

引用

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

作者： S. Meister R. Wiggins Electronic Systems Division Hanscom AFB Ma. MITRE Corporation Bedford MA USA

This paper reports the results of an investigation of a computable Quality Comparison Measure (called the QCM) for linear predictive systems. The measure described is easily obtained by a synthesis-analysis procedure. It is a weighted combination of differences between the input and output speech parameters for a series of spoken sentences. Results are presented that demonstrate a high correlation between QCM and listener preference scores. The QCM offers an alternative to costly and time consuming formal listening procedures.

关键词： Reflection linear predictive coding Repeaters Speech analysis Equations Distortion measurement Speech synthesis Speech coding System testing Synthesizers

来源：评论

学校读者我要写书评

暂无评论

On reducing the bit rate of a CELP-based speech coder

On reducing the bit rate of a CELP-based speech coder

引用

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

作者： Y.J. Liu Communications Division ITT-Aerospace Nutley NJ USA

For a code excited linear prediction (CELP)-based coder, coarse quantization usually results in the degradation of speech quality. Spectrum and gain were found to have the least sensitivity to quantization errors and were efficiently quantized. The frame period and number of subframes per frame were also appropriately selected for the purpose of bit-rate reduction. The performance of four possible excitations, namely, stochastic excitation, binary pulse excitation, self-excitation and repeat excitation were investigated. Using signal-to-noise ratio as a quality measure, the coder at 3076 b/s was found to have a performance very close to that of the federal standard at 4800 b/s.< >

关键词： Bit rate Quantization Speech coding linear predictive coding Degradation Speech analysis Stochastic processes Speech synthesis predictive models Distortion measurement

来源：评论

学校读者我要写书评

暂无评论

Hybrid audio coding for speech and audio below medium bit rate

Hybrid audio coding for speech and audio below medium bit ra...

引用

IEEE International Conference on Consumer Electronics (ICCE)

作者： K. Makino J. Matsumoto Dev. Center Sony Corp. Tokyo Japan HomeNet Processing Laboratory Development CenterHome Network Company Sony Corporation USA

A novel a audio/speech coding algorithm, hybrid audio coding (HAC) is described. New features of the algorithm include window switching with generalized MDCT, an improved quantization scheme of the MDCT coefficients, and waveform normalization in the time domain. HAC provides a good quality at a bit rate of 8 to 16 kbps, and it is also proven that the developed algorithm is effective for both audio and speech signals.

关键词： Audio coding Speech coding Bit rate Codecs Testing linear predictive coding Quantization Multiple signal classification Decoding Smoothing methods

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 << < 224 225 226 227 228 229 230 231 232 233 > >>

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：