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

268 篇 会议
10 篇 期刊文献

馆藏范围

278 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

119 篇 工学
- 72 篇 信息与通信工程
- 36 篇 计算机科学与技术...
- 30 篇 软件工程
- 29 篇 仪器科学与技术
- 26 篇 电子科学与技术（可...
- 17 篇 电气工程
- 9 篇 控制科学与工程
- 8 篇 生物医学工程（可授...
- 8 篇 生物工程
- 3 篇 机械工程
- 3 篇 建筑学
- 2 篇 光学工程
- 2 篇 土木工程
- 2 篇 交通运输工程
- 1 篇 动力工程及工程热...
- 1 篇 化学工程与技术
67 篇 理学
- 39 篇 物理学
- 23 篇 数学
- 20 篇 系统科学
- 8 篇 生物学
- 8 篇 统计学（可授理学、...
- 1 篇 化学
7 篇 医学
- 7 篇 临床医学
- 2 篇 基础医学(可授医学...
- 1 篇 药学(可授医学、理...
5 篇 管理学
- 5 篇 管理科学与工程(可...
- 1 篇 工商管理
- 1 篇 图书情报与档案管...
3 篇 文学
- 3 篇 新闻传播学
2 篇 法学
- 2 篇 社会学
1 篇 军事学

主题

44 篇 decoding
41 篇 speech
35 篇 data processing
34 篇 iterative decodi...
31 篇 speech enhanceme...
30 篇 codecs
24 篇 signal to noise ...
23 篇 speech coding
21 篇 redundancy
21 篇 bit rate
19 篇 channel coding
18 篇 bandwidth
18 篇 delay
17 篇 noise reduction
17 篇 noise
16 篇 speech processin...
15 篇 computational co...
14 篇 speech codecs
14 篇 microphones
14 篇 wideband

机构

117 篇 institute of com...
31 篇 institute of com...
22 篇 institute of com...
14 篇 institute of com...
9 篇 institute of com...
9 篇 institute of com...
7 篇 institute for co...
6 篇 institute of inf...
5 篇 institute of com...
5 篇 rheinisch-westfa...
4 篇 institute for co...
4 篇 institute for in...
3 篇 institute of com...
3 篇 institute for co...
3 篇 intel mobile com...
3 篇 institute of com...
3 篇 institute of com...
3 篇 fkie communicati...
3 篇 institute of com...
2 篇 institute of com...

作者

123 篇 peter vary
49 篇 p. vary
44 篇 vary peter
17 篇 bernd geiser
15 篇 laurent schmalen
12 篇 heinrich w. loll...
12 篇 m. adrat
11 篇 thorsten clevorn
10 篇 t. clevorn
10 篇 birgit schotsch
9 篇 marco jeub
8 篇 tobias bredderma...
8 篇 christophe beaug...
7 篇 marc adrat
7 篇 thomas esch
7 篇 p. jax
7 篇 löllmann heinric...
6 篇 krüger hauke
6 篇 moritz beermann
6 篇 s. heinen

语言

270 篇 英文
8 篇 中文

检索条件"机构=Institute of Communication Systems and Data Processing RWTH Aachen University"

共 278 条记录，以下是141-150 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：

Hybrid ARQ with incremental redundant index assignments for iterative source-channel decoding

Hybrid ARQ with incremental redundant index assignments for ...

引用

2010 IEEE International Conference on communications, ICC 2010

作者： Clevorn, Thorsten Schmalen, Laurent Vary, Peter Adrat, Marc Design Center NRW Infineon Technologies AG Duisburg Germany Inst. of Comm. Systems and Data Processing RWTH Aachen University Aachen Germany FKIE Communication Systems Fraunhofer-Gesellschaft Wachtberg Germany

ISBN: (纸本)9781424464043

Iterative source-channel decoding (ISCD) exploits the residual redundancy of source codec parameters by using the Turbo principle. In this paper we extend the excellent capabilities of ISCD to a hybrid automatic repeat request (HARQ) scheme by novel incremental redundant index assignments. The incremental redundancy for HARQ is supplied by the source encoder and not the channel encoder. Simulation results show an excellent performance over a wide range of channel conditions, with inherently adapted bandwidth and complexity. ©2010 IEEE.

关键词： Signal encoding

来源：评论

学校读者我要写书评

暂无评论

Logarithmic Cubic Vector Quantization: Concept and analysis

Logarithmic Cubic Vector Quantization: Concept and analysis

引用

International Symposium on Information Theory and Its Applications, ISITA

作者： Christian Rohlfing Hauke Krüger Peter Vary Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University Germany

In this paper, we analyze Logarithmic Cubic Vector Quantization (LCVQ), a novel type of gain-shape vector quantization (GSVQ). In LCVQ, the vector to be quantized is decomposed into a gain factor and a shape vector which is a normalized version of the input vector. Both components are quantized independently and transmitted to the decoder. Compared to other GSVQ approaches, in LCVQ the input vectors are normalized based on the maximum norm (also denoted as L ∞ -norm) instead of the typically used Euclidean norm (L 2 -norm). Therefore, all shape vectors are located on the surface of the unit hypercube. As a conclusion, the shape vector quantizer can be realized based on uniform scalar quantizers yielding low computational complexity as well as high memory efficiency even in case of very high vector dimensions. In this paper, the concept of LCVQ is presented. Also, theoretical quantization performance measures for LCVQ as well as the optimal allocation of bit rate for gain factor and shape vector are derived. In order to assess the proposed LCVQ approach, the quantization performance achieved by LCVQ is compared to results which were recently derived for Logarithmic Spherical Vector Quantization (LSVQ), another highly efficient GSVQ scheme proposed in.

关键词： Vectors Signal to noise ratio Bit rate Shape Vector quantization Equations

来源：评论

学校读者我要写书评

暂无评论

On logarithmic spherical vector quantization

On logarithmic spherical vector quantization

引用

International Symposium on Information Theory and Its Applications, ISITA

作者： Hauke Kruger Raimund Schreiber Bernd Geiser Peter Vary Institute of Communication Systems and Data Processing (IND) RWTH Aachen Aachen Germany

ISBN: (纸本)9781424420681

Logarithmic spherical vector quantization (LSVQ) is a specific type of gain-shape vector quantization (VQ), where input vectors are decomposed into a gain and a shape component which are quantized independently. In this contribution, novel theoretical results on LSVQ are presented: It will be shown that, for high bit rates, with logarithmic (A-Law) scalar quantization (SQ) of the gain and spherical vector quantization (SVQ) of the shape component a signal-to-noise ratio (SNR) is achieved which is approximately independent of the input source distribution. In addition, a detailed theoretical analysis leads to a lower bound for the quantization distortion related to SVQ. By introducing approximations for the assumption of high bit rates, this bound is the basis for the computation of the optimal allocation of bit rate to the gain and the shape quantizer, respectively, and yields an estimate for the achievable SNR for LSVQ.

关键词： Vector quantization Shape Bit rate Computational complexity Dynamic range Information theory communication systems data processing Signal to noise ratio Independent component analysis

来源：评论

学校读者我要写书评

暂无评论

Measurement, analysis and simulation of wind noise signals for mobile communication devices

Measurement, analysis and simulation of wind noise signals f...

引用

International Workshop on Acoustic Signal Enhancement (IWAENC)

作者： Christoph Matthias Nelke Peter Vary Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University Germany

In this contribution, we study the characteristics of sound generated by wind and a signal model for the synthesis of wind noise signals is derived. An analysis of the statistics of wind noise recorded in a laboratory setup is carried out with respect to the spectral and temporal properties of the signals. In particular, an autoregresive model is developed for the spectral shape description and the temporal statistics are modeled by a Markov chain. These two components are combined in a model which synthesizes reproducible artificial wind noise signals. Furthermore, a database of measured wind noise signals is provided. The aim of this model and the measured audio data is to provide wind signals for the evaluation of speech enhancement and noise reduction systems.

关键词： Noise Speech Noise measurement Microphones Acoustics databases Mathematical model

来源：评论

学校读者我要写书评

暂无评论

Surpassing Purely Digital Transmission: A Simplified Design of Hybrid Digital Analog Codes

Surpassing Purely Digital Transmission: A Simplified Design ...

引用

SCC 2013;9th International ITG Conference on systems, communication and Coding

作者： Matthias Ruengeler Peter Vary Institute of Communication Systems and Data Processing (ind) RWTH Aachen University Germany

One problem of conventional digital transmission systems that transmit real-valued source symbols is that the transmission fidelity always saturates for channels that have attained the design channel quality (worst-case channel condition); this saturation is caused by the irreversible error introduced by the source coder. Even though hybrid digital analog (HDA) codes address this problem by additionally transmitting the inherent quantization error by using analog methods, the design and the decoding of these HDA codes is complex or even impossible for long block lengths. Thus, the aim of this study is to introduce a new way to design HDA codes using well-known digital codes. Another objective is to theoretically and empirically compare the proposed design with the conventional purely digital transmission systems. This study applied Monte Carlo simulations with Reed- Solomon codes or convolutional codes as the digital part and LMMSE estimation in the analog part of the HDA system. The simulation results agreed well with the theoretical considerations. Since the proposed design uses well-established, excellent digital codes that support long block lengths, the design and decoding of HDA codes is simplified. The inherent error introduced by the source coder is reduced by combining these digital codes with analog transmission, thereby eliminating the saturation of the transmission fidelity. The newly proposed HDA system surpasses the purely digital transmission system at all channel qualities.

关键词： Decoding Quantization Digital systems PSNR Vectors Channel coding

来源：评论

学校读者我要写书评

暂无评论

Iterative source-channel decoder using extrinsic information from softbit-source decoding

Iterative source-channel decoder using extrinsic information...

引用

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

作者： M. Adrat P. Vary J. Spittka Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Germany

In digital mobile communications, efficient compression algorithms are needed to encode speech or audio signals. As the determined source parameters are highly sensitive to transmission errors, robust source and channel decoding schemes are required. This contribution deals with an iterative source-channel decoding approach where a simple channel decoder and a softbit-source decoder are concatenated. We mainly focus on softbit-source decoding which can be considered as an error concealment technique. This technique utilizes residual redundancy remaining after source coding. We derive a new formula that shows how the residual redundancy transforms into extrinsic information utilizable for iterative decoding. The derived formula opens several starting points for optimizations, e.g. it helps to find a robust index assignment. Furthermore, it allows the conclusion that softbit-source decoding is the limiting factor if applied to iterative decoding processes. Therefore, no significant gain will be obtainable by more than two iterations. This will be demonstrated by simulation.

关键词： Iterative decoding Redundancy Iterative algorithms Parameter estimation Robustness Iterative methods Source coding Channel coding data processing Mobile communication

来源：评论

学校读者我要写书评

暂无评论

Near-End Listening Enhancement in the Presence of Bandpass Noises

Near-End Listening Enhancement in the Presence of Bandpass N...

引用

Speech communication;10. ITG Symposium

作者： Bastian Sauert Peter Vary Institute of Communication Systems and Data Processing (i~d) RWTH Aachen University

ISBN: (纸本)9783800734559

When using a mobile phone in acoustical background noise, the near-end listener perceives not only the (clean) far-end speech but also the ambient noise and thus experiences an increased listening effort and a possibly reduced speech intelligibility. Near-end listening enhancement processes the received far-end speech signal depending on the near-end background noise to improve intelligibility. However, in mobile phones it is often not possible to increase the audio power. In a previous contribution, the authors developed a recursive closed-form solution, which maximizes the Speech Intelligibility Index (SII) under the constraint of an unchanged average power of the audio signal. This solution, however, shows in bandpass noise environments a disadvantageous narrow bandpass characteristic of the processed speech even though the SII is optimal. Therefore, in this contribution, we analyze this algorithm and propose a new spectral weighting rule which prevents the narrow bandpass effect with only a marginal reduction in SII.

关键词： Speech Signal to noise ratio Optimization Indexes Speech enhancement Noise measurement

来源：评论

学校读者我要写书评

暂无评论

HYBRID DIGITAL ANALOG TRANSFORM CODING

HYBRID DIGITAL ANALOG TRANSFORM CODING

引用

IEEE International Conference on Acoustics, Speech and Signal processing

作者： Matthias Rungeler Peter Vary Institute of Communication Systems and Data Processing (ind) RWTH Aachen University Germany

ISBN: (纸本)9781479903573

One problem of conventional digital transmission systems is that the transmission fidelity always saturates at the design channel quality (worst-case channel condition);this saturation is due to the irreversible quantization error introduced by the source coder. Hybrid digital analog (HDA) codes address this problem by additionally transmitting the inherent quantization error by pseudo-analog methods. However, the design and the decoding of these HDA codes is complex or even impossible for long block lengths (look-up table decoding). In this study we propose an HDA transmission system for sources with correlation supporting long block lengths. This is achieved by combining transform coding with HDA transmission using well-known digital channel codes. For the evaluation Monte Carlo simulations are used with a normalized discrete cosine transform (DCTN) and Turbo codes as the digital part and LMMSE estimation in the pseudo-analog part of the HDA system. The proposed HDA system with transform coding outperforms the purely digital transmission system at all channel qualities and exploits the expected gain due to source correlation.

关键词： Hybrid Digital Analog (HDA) transform coding Turbo code LMMSE estimator

来源：评论

学校读者我要写书评

暂无评论

Improved Binaural Model for Localization of Multiple Sources

Improved Binaural Model for Localization of Multiple Sources

引用

Speech communication;10. ITG Symposium

作者： Magnus Schaefer Mohammad Bahram Peter Vary Institute of Communication Systems and Data Processing (i~d) RWTH Aachen University

ISBN: (纸本)9783800734559

An improved binaural hearing model is proposed that consists of a physiologically motivated signal processing step and a subsequent cognitive model. The model is shown to be capable of correctly detecting and tracking sources while blindly determining the number of active sources based on temporal and spectral information. The complete model is of interest in all areas that need to consider the capabilities of the human hearing system while the ability to determine the number of active sources makes it a logical enhancement for source separation and spatial signal processing such as adaptive beamforming.

关键词： Humans Psychoacoustic models Mathematical model Ear Skeleton Brain modeling

来源：评论

学校读者我要写书评

暂无评论

Speech bandwidth extension based on in-band transmission of higher frequencies

Speech bandwidth extension based on in-band transmission of ...

引用

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

作者： Bernd Geiser Peter Vary Institute of Communication Systems and Data Processing (ind) RWTH Aachen University Germany

ISBN: (纸本)9781479903573

A new method for wideband speech transmission is proposed which is fully backwards compatible with narrowband telephone systems. For this purpose, a pitch-scaled version of the higher speech frequencies (4 - 6.4 kHz) is inserted into the previously “unused” 3.4 - 4 kHz frequency range of standard telephone speech. This operation is reverted at the decoder side. A consistently good wideband speech quality can be achieved, even after transmission over common codecs and codec tandems. The quality impact on the narrowband part of the signal is insignificant.

关键词： Speech Narrowband Wideband Codecs Speech coding Speech processing

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28页 << < 11 12 13 14 15 16 17 18 19 20 > >>

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：