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
406 篇 期刊文献
4 篇 学位论文
1 册 图书

馆藏范围

2,457 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

601 篇 工学
- 442 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

2,457 篇 linear predictiv...
770 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

2,283 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 2457 条记录，以下是861-870 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：

Novel dynamic bit allocation method for LSP quantization

Novel dynamic bit allocation method for LSP quantization

引用

IEEE Region 10 International Conference TENCON

作者： Fang Hui I. McLoughlin School of Computer Engineering Nanyang Technological University Singapore

The line spectral pair (LSP) conveys vocal tract information for reconstruction of speech in an analysis-synthesis speech coding system. Quantization of LSP parameters is necessary to reduce bit-rate, but causes unavoidable degradation in quality. This paper proposes an adaptive quantization method to reduce bit-rate whilst maintaining reasonable speech quality. The method exploits the relationship between LSP position and speech features.

关键词： Bit rate Quantization Speech analysis linear predictive coding Speech coding Frequency domain analysis Information analysis Degradation Spectral analysis Robustness

来源：评论

学校读者我要写书评

暂无评论

A new algorithm for parameter re-optimization in multi-pulse excitation LP synthesizer

A new algorithm for parameter re-optimization in multi-pulse...

引用

IEEE Asia-Pacific Conference on Circuits and Systems

作者： Yining Chen Penghao Wang Jia Liu Runsheng Liu Department of Electronic Engineeringy Tsinghua University Beijing China

ISBN: (纸本)0780362535

A new algorithm is proposed to re-optimize the synthesis filter parameters and pulse amplitudes in the Multi-Pulse Excitation linear Prediction (LP) synthesizer. It is based on minimizing the perceptional weighted mean square errors between the original and the reconstructed speech. Steepest gradient descent and conjugate gradient descent are employed. The LP filter parameters and all the pulse amplitudes are iteratively re-optimized. Compared with the conventional multi-pulse excitation LPC algorithm, the new algorithm improves the segmental signal-to-noise ratio (SEGSNR) from about 20 dB to 23 dB. The informal listening also shows that the quality of synthesized speech is improved greatly.

关键词： Synthesizers Speech synthesis Equations Mean square error methods Iterative algorithms Signal synthesis Autocorrelation Electronic mail Nonlinear filters linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

Feature windowing-based Thai text-dependent speaker identification using MLP with backpropagation algorithm

Feature windowing-based Thai text-dependent speaker identifi...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Sutat Sae-Tang C. Tanprasert National Science and Technology Development Agency National Electronics and Computer Technology Center Software and LanguagJ Engineering Laboratory Ministry of Science Technology and Environment Bangkok Thailand

This paper presents the development of Thai text dependent speaker identification system by applying two feature-feeding approaches. A well-known multilayer perceptron (MLP) network with backpropagation learning algorithm is chosen. It has fast processing time and good performance for pattern recognition problems. But MLP has a limitation in that a network must have a fixed amount of input nodes. Therefore, the linear interpolation time normalization is chosen to adjust the input speech signal into a fixed size of input vector. Furthermore, the windowing technique is developed to avoid the distortion caused by a time normalization process. A fixed size window is sliced through the preprocessed features with fixed amount of overlapping frames. The high identification rate observed in experiments confirms that the developed windowing is suitable for the proposed Thai text-dependent speaker identification system.

关键词： Speaker recognition Backpropagation algorithms Multilayer perceptrons Natural languages Feature extraction Speech recognition Engines linear predictive coding Cepstral analysis Frequency

来源：评论

学校读者我要写书评

暂无评论

Variable rate multi-mode excitation coding of speech at 2.4 kbps

Variable rate multi-mode excitation coding of speech at 2.4 ...

引用

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

作者： Shihua Wang Atmel Corporation Berkeley CA USA

A variable rate speech coder utilizing speech classification is presented as a cost effective solution to applications where low bit rate, good quality, and less complexity are the main objectives. The coder operates at an average rate of 2.4 kbps and the speech quality is close to that produced by the ITU G.723.1 codec. It has been implemented with less than 10 MIPS, 2.5 K RAM, 3.3 K DROM, and 6 K PROM on a 16-bit fixed-point general purpose DSP. The framework of the coder is based on multi-pulse and code-excited coding schemes. The rate variation and coding efficiency are manifested in a multimode excitation modeling depending on the nature of the input speech.

关键词： Speech coding linear predictive coding Bit rate Costs Code standards Material storage Speech codecs PROM Digital signal processing Telephony

来源：评论

学校读者我要写书评

暂无评论

Speech rate conversion without extraction of accurate pitch period

Speech rate conversion without extraction of accurate pitch ...

引用

Annual Conference of Industrial Electronics Society

作者： A. Shiokawa K. Yamada R. Ishii Division of Electrical and Computer Engineering Yokohama National University Yokohama Japan Yokohama Kokuritsu Daigaku Yokohama Kanagawa JP

Many speech rate conversion methods have been proposed. In a speech rate conversion, there are two kinds of processes: one is a contraction of a speech period and the other is an extension. Specially, the extension of speech period is useful for aged persons to listen easily without changing its composed frequencies of speech and personality. In these methods, the extraction of an accurate pitch period in a voice is needed. However, an accurate pitch period extraction is very complicated. This paper proposes a speech rate conversion without the extraction of an accurate pitch period. As a result, the constructed speech by the proposed method gives a better quality than other convention methods, according to the calculation of power weighted LPC cepstrum distance and the percent error of its output sample numbers.

关键词： Speech processing Tellurium Aging Teleprinting linear predictive coding Cepstrum Frequency Autocorrelation

来源：评论

学校读者我要写书评

暂无评论

Pitch detection of speech signals using the cross-correlation technique

Pitch detection of speech signals using the cross-correlatio...

引用

IEEE Region 10 International Conference TENCON

作者： S.A. Samad A. Hussain Low Kok Fah Multimedia Signal Processing Research Group Department of Electrical Electronic and Systems Engineering Faculty of Engineering Universiti Kebangsaan Malaysia Selangor Malaysia

To generate high quality speech using the linear predictive coding (LPC) technique, a method for detecting pitch contour is critical since the human ear is sensitive to small pitch variation in speech. The auto-correlation method, though simple to implement with digital signal processors (DSPs), can result in perceptible unnaturalness. This paper describes the cross-correlation technique that can be used to obtain the pitch information more accurately than the auto-correlation method for certain speech samples. Experimental results illustrate the pitch contour detected using both techniques. In general, the cross-correlation method generates less error than the auto-correlation method for pitch determination in an LPC scheme while having the advantage of requiring less computation.

关键词： Signal detection Autocorrelation linear predictive coding Speech processing Correlation Delay Filters Speech synthesis Speech coding Humans

来源：评论

学校读者我要写书评

暂无评论

A sinusoidal LPC vocoder

A sinusoidal LPC vocoder

引用

IEEE Workshop on Speech coding Proceedings

作者： P. Hedelin Department of Signals and Systems Information Theory Group Gothenburg Sweden

Twenty years of work with sinusoidal modeling of speech has lead to very competitive principles of low rate coding. In this study, we discuss a few issues in the design of a sinusoidal coding system. We stress that by a careful design of all blocks of the encoder and decoder, allowing for some additional complexity, it is possible to build a low rate coder free of many of the artifacts associated with conventional vocoding systems. We focus this discussion on multi-band partial voicing and on parameter smoothing and interpolation.

关键词： linear predictive coding Vocoders Speech coding Power harmonic filters Signal generators Interpolation Speech processing Filtering Speech enhancement Filter bank

来源：评论

学校读者我要写书评

暂无评论

Implementation of ITU-T G.723.1 dual rate speech codec based on TMS320C6201 DSP

Implementation of ITU-T G.723.1 dual rate speech codec based...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Yongfeng Huang Yang Juan Sifa Zhang Jiangling Zhang National Storage System Laboratory School of Computer Huazhong University of Science and Technology Wuhan China

This paper describes how to implement the G.723.1 recommendation in the IP telephony gateway and studies in detail the programming of the TMS320C6201 DSP and optimization methods for reducing the speech processing delay of the G.723.1 codec. As a result of adopting these optimization and programming methods, we have implemented a high-speed speech codec which can process concurrently 18 voice channels with a single TMS320C6201 chip in the IP telephony gateway. Finally, the paper summarizes the performance of the resulting ITU-T G.723.1 speech codec.

关键词： Speech codecs Digital signal processing Internet telephony Filters linear predictive coding Decoding Optimization methods Speech processing Delay effects Speech coding

来源：评论

学校读者我要写书评

暂无评论

The effect of polarity inversion of speech on human perception and data hiding as an application

The effect of polarity inversion of speech on human percepti...

引用

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

作者： S. Sakaguchi T. Arai Y. Murahara Department of Electrical and Electronics Engineering Sophia University Chiyoda Tokyo Japan

In this paper we investigate how polarity inversion of speech signals effects human perception, and we apply this technique for data hiding. In most languages, glottal airflow during phonation is uni-directional, causing constant polarity of the speech waveform. On the other hand, the human auditory system cannot discriminate between speech signals with positive and negative polarity. Based on these facts, we developed an algorithm to hide data in speech signals. We assigned one bit to each syllable of speech, and inverted the polarity of the signal at every syllable according to the assigned bit. We performed a test using 20 sentences from the TIMIT corpus to determine both whether a human could distinguish between the original and polarity-inverted signal and whether we could automatically restore the embedded binary data. We found that we were able to successfully hide data and restore it automatically.

关键词： Humans Data encapsulation linear predictive coding Speech analysis Auditory system Signal restoration Filtering Signal analysis Filters Frequency

来源：评论

学校读者我要写书评

暂无评论

Calculation of the line spectrum frequencies using the quotient-difference scheme

Calculation of the line spectrum frequencies using the quoti...

引用

Mediterranean Electrotechnical Conference (MELECON)

作者： D. Petrinovic Department of Electronic Systems and Information Processing Faculty of Electrical Engineering and Computing University of Zagreb Croatia

A new, computationally efficient technique for calculation of the line spectrum frequencies (LSF) that can be applied to any order of the LPC analysis is proposed. It is based on the quotient-difference (Q-D) root-finding algorithm that enables simultaneous solution for all the LSFs. It is an iterative procedure that offers the tradeoff between accuracy and complexity, what is especially important for the real-lime applications. To improve the convergence, a nonlinear mapping of the LSFs is also proposed for low accuracy applications, the method is even more effective then the fast converging Newton-Rapshon method, but is at the same time exceptionally simple, has a very regular structure and requires only basic mathematical operations.

关键词： Frequency linear predictive coding Polynomials Speech Equations Convergence Iterative algorithms Filters Chebyshev approximation Electronic equipment testing

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 << < 83 84 85 86 87 88 89 90 91 92 > >>

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：