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171-180 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：

Efficient non-uniform filter-bank equalizer

Efficient non-uniform filter-bank equalizer

引用

European Signal processing Conference (EUSIPCO)

作者： Heinrich W. Löllmann Peter Vary Institute of Communication Systems and Data Processing (i~d) RWTH Aachen University Aachen Germany

An efficient realization of a low delay filter-bank will be proposed, which can be regarded as a filter-bank used as equalizer with either fixed or time-varying spectral gain factors. The proposed filter-bank equalizer (FBE) results in a time-domain filter whose coefficients are adapted in the (short-term) spectral-domain. Perfect signal reconstruction is obtained for a broad class of spectral transforms, including the generalized discrete Fourier transform (GDFT), the Walsh and Hadamard transform, with less restrictions compared to a common analysis-synthesis filter-bank (AS FB). A non-uniform frequency resolution can be achieved by frequency warping based on an allpass transformation. In this case, the filter-bank equalizer can achieve near perfect signal reconstruction with lower effort than for an allpass transformed analysis-synthesis filter-bank. The filterbank equalizer is especially useful for adaptive filtering requiring a low signal delay with coefficient adaptation based on frequency domain algorithms.

关键词： Equalizers Delays Discrete Fourier transforms Complexity theory Prototypes Noise reduction

来源：评论

学校读者我要写书评

暂无评论

NEAR END LISTENING ENHANCEMENT OPTIMIZED WITH RESPECT TO SPEECH INTELLIGIBILITY INDEX

NEAR END LISTENING ENHANCEMENT OPTIMIZED WITH RESPECT TO SPE...

引用

European signal processing conference

作者： Bastian Sauert Peter Vary Institute of Communication Systems and Data Processing (ind) RWTH Aachen University 52056 Aachen Germany

ISBN: (纸本)9781617388767

Signal processing algorithms for near end listening enhancement allow to improve the intelligibility of clean (far end) speech for the near end listener who perceives not only the far end speech but also ambient background noise. A typical scenario is mobile communication conducted in the presence of acoustical background noise such as traffic or babble noise. In this contribution we analyze the calculation rules of the Speech Intelligibility Index (SII) and derive a simple condition for the speech spectrum level of every subband that maximizes the SII for a given noise spectrum level. This rule is used to derive a theoretical bound for a maximum achievable SII as well as a new SII optimized algorithm for near end listening enhancement. The impact of ignoring masking effects in the algorithm is also investigated and seconds our SNR recovery algorithm proposed earlier. Instrumental evaluation shows that the new algorithm performs close to the established theoretical bound.

关键词： intelligibility Speech BACKGROUND NOISE Signal processing algorithms algorithms mobile communication Traffic

来源：评论

学校读者我要写书评

暂无评论

A soft-partitioned frequency-domain adaptive filter for acoustic echo cancellation

A soft-partitioned frequency-domain adaptive filter for acou...

引用

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

作者： G. Enzner P. Vary Institute of Communication Systems and Data Processing (IND) RWTH Aachen University of Technology Aachen Germany

We present a soft-partitioned frequency-domain adaptive filter (SPFDAF) and compare it in the context of other LMS type acoustic echo cancelers. The SPFDAF uses a nonrectangular window to partition the model taps (impulse response) of an echo canceler. Our soft window can be approximated very efficiently in the DFT domain and still attenuates aliasing components of the DFT by 40 dB. This technique results in a very low computational complexity and maintains an excellent signal quality.

关键词： Adaptive filters Echo cancellers Frequency Least squares approximation Convergence Loudspeakers Computational complexity Partitioning algorithms Delay data processing

来源：评论

学校读者我要写书评

暂无评论

GENERAL LEAST-SQUARES DESIGN OF ALLPASS TRANSFORMED DFT FILTER-BANKS

GENERAL LEAST-SQUARES DESIGN OF ALLPASS TRANSFORMED DFT FILT...

引用

European signal processing conference

作者： Heinrich W. Lollmann Guido Dartmann Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany now with Institute for Integrated Signal Processing Systems RWTH Aachen University 52056 Aachen Germany

ISBN: (纸本)9781617388767

The allpass transformation is a common approach to design a frequency warped analysis-synthesis filter-bank (AS FB) with non-uniform time-frequency resolution. Such filter-banks are beneficial, e.g., for speech and audio subband processing, but their synthesis filter-bank design is more complex as for uniform filter-banks. This problem is tackled by a new least-squares error (LS) synthesis filter-bank design. Magnitude and phase distortions are minimized with the constraints for complete aliasing cancellation and a restricted signal delay. The coefficients of the synthesis filters are determined by a Jinear set of equations with linear constraints which can be easily solved. The obtained FIR synthesis subband Slters exhibit a frequency selectivity similar to that of the IIR analysis filters. The new concept is very general and can be applied in case of an allpass transformation of first or higher order as well as for polyphase network (PPN) filter-banks where the prototype filter length exceeds the number of channels. These properties make the proposed design of interest for various subband processing systems requiring non-uniform frequency bands.

关键词： synthesis filter prototype filter subband Phase distortion frequency selectivity synthesis Frequency bands Filter banks Filters

来源：评论

学校读者我要写书评

暂无评论

Iterative source-channel decoding with cross-layer support for wireless VoIP

Iterative source-channel decoding with cross-layer support f...

引用

International ITG Conference on Source and Channel Coding (SCC)

作者： Tobias Breddermann Helge Lüders Peter Vary Ismet Aktas Florian Schmidt Institute of Communication Systems and Data Processing (iNti) RWTH Aachen University Germany Distributed Systems Group RWTH Aachen University Germany

This paper presents a cross-layer approach for iterative source-channel decoding (ISCD) in wireless VoIP networks. The novelty of the proposed method is the incorporation of both, speech bits as well as protocol header bits, into the ISCD process. The header bits take the role of pilot bits having perfect reliability. These bits are distributed over the frame as strong supporting points for the MAP decoder which. results in a significant enhancement of the output speech quality compared to the benchmark scheme using ISCD for speech only. For this approach, we exploit cross-layer concepts that support the direct communication between non-adjacent layers. These concepts enable the iterative exchange of extrinsic information between the source decoder located on the application layer and the channel decoder located on the physical layer. This technique can also be applied to audio and video transmission.

关键词： Iterative decoding Speech processing Redundancy Iterative methods Speech enhancement Physical layer Bandwidth Internet telephony Open systems data processing

来源：评论

学校读者我要写书评

暂无评论

Iterative source coded modulation: Exit charts, complexity comparisons and new index assignments

Iterative source coded modulation: Exit charts, complexity c...

引用

European Signal processing Conference (EUSIPCO)

作者： Thorsten Clevorn Peter Vary Institute of Communication Systems and Data Processing (i~d) RWTH Aachen University Germany

Iterative source coded modulation (ISCM) improves the error concealment for source codec parameters without increasing the transmitted bit rate by combining iterative demodulation of higher order modulations and the usage of residual source redundancy in a Turbo process. In this paper we present the enhanced capabilities of ISCM when novel index assignments are applied. A doubling of the previously achievable logarithmic gains [1] is possible. Furthermore, the convergence properties of ISCM are analyzed by EXIT charts and the computational complexity is compared to a rate-1 iterative source-channel decoding (ISCD) system [2]. ISCM exhibits a competitive performance in relation to an equally complex ISCD system.

关键词： Indexes Niobium Modulation Complexity theory Signal to noise ratio Abstracts AWGN

来源：评论

学校读者我要写书评

暂无评论

ROBUST DUAL-CHANNEL NOISE POWER SPECTRAL DENSITY ESTIMATION

ROBUST DUAL-CHANNEL NOISE POWER SPECTRAL DENSITY ESTIMATION

引用

European Signal processing Conference

作者： Marco Jeub Christoph Nelke Hauke Kruger Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Intel Mobile Communications

In this paper, we propose a novel noise power spectral density (PSD) estimator which is beneficial for speech enhancement systems with two microphones in diffuse noise environments. The algorithm has a low computational complexity and requires low memory usage. The main advantage is that arbitrary models of the noise field coherence can be employed and a scalable extension of existing single-channel speech enhancement systems to dual channels is also possible. Experiments demonstrate with simulated and measured data that the proposed algorithm outperforms related algorithms in diffuse noise conditions.

关键词： noise field Spectral density Speech enhancement channel noise Noise noise environment noise power spectral density algorithms complexity classes observational data

来源：评论

学校读者我要写书评

暂无评论

Quality control for AMR speech channels in GSM networks

Quality control for AMR speech channels in GSM networks

引用

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

作者： M. Werner T. Junge P. Vary Institute of Communication Systems and Data Processing(IND) RWTH Aachen University of Technology Germany

In the competitive environment of cellular radio networks, the speech quality experienced by the end user is of vital importance for the network operator. Therefore, the operator needs reliable measures for monitoring and controlling the speech quality. For GSM mobile radio networks employing the adaptive multi-rate (AMR) speech codec, a new instrumental and non-intrusive speech quality metric is proposed. 'Non-intrusive' means that the metric is based only on received transmission parameters, such as the frame erasure rate, and needs neither the original nor the transmitted speech samples. However, for the validation of a new metric, standardized intrusive speech quality measures like PESQ, link-level simulations and field measurements are used. Furthermore, for an improved control of speech quality, a novel AMR mode switching procedure is introduced which outperforms the recommended method of the GSM standard.

关键词： Quality control Intelligent networks GSM Speech codecs Testing Distortion measurement Receivers Bit error rate Monitoring Land mobile radio

来源：评论

学校读者我要写书评

暂无评论

BLIND ESTIMATION OF THE COHERENT-TO-DIFFUSE ENERGY RATIO FROM NOISY SPEECH SIGNALS

BLIND ESTIMATION OF THE COHERENT-TO-DIFFUSE ENERGY RATIO FRO...

引用

European Signal processing Conference

作者： Marco Jeub Christoph Nelke Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Intel Mobile Communications

A novel dual-channel algorithm is proposed which estimates the coherent-to-diffuse energy ratio (CDR) of background noise in mixed noise fields. The algorithm is based on an estimate of the noise field coherence from a noisy speech signal and a subsequent minima tracking in order to increase the estimation accuracy even in the presence of speech. The obtained CDR estimate can be used, e.g., for the acoustic environment classification in hearing aids or to control speech enhancement algorithms such as noise reduction or speech dereverberation. Besides, the approach can be used to calculate an estimate of the direct-to-reverberant energy ratio (DRR) blindly from reverberant speech signals.

关键词： Noise Speech Coherence Estimation Noise measurement Frequency estimation Signal processing algorithms

来源：评论

学校读者我要写书评

暂无评论

Robust equalizer design for allpass transformed DFT filter banks with LTI property

Robust equalizer design for allpass transformed DFT filter b...

引用

IEEE International Symposium on Personal, Indoor and Mobile Radio communications (PIMRC)

作者： Fabian Altenbach Heinrich W. Löllmann Rudolf Mathar Institute for Theoretical Information Technology RWTH Aachen University Aachen Germany Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany

Allpass transformed filter banks provide a nonuniform frequency resolution and can be used in mobile speech processing systems, e.g., cellular phones or digital hearing aids. The nominal design of such an allpass transformed analysis-synthesis filter bank (AS FB) with near perfect reconstruction (NPR) is achieved by numerical optimization of finite-impulse response (FIR) equalizers in each subchannel. The underlying nominal optimization problem is an equality constrained leasts-quares problem. In a robust design, we take into account coefficient uncertainty in a possible implementation of such a filter bank. We will describe this uncertainty by the choice of two simple set-based worst-case uncertainty models, namely a norm bound error model and a coefficient bound error model. When including these error models, both robust designs can be recast as second-order cone programs (SOCP) and solved efficiently by standard numerical optimization methods. Furthermore, we will provide design examples to show that both robust designs maintain a good overall performance with respect to NPR while offering less sensitivity to quantization errors.

关键词： Robustness Uncertainty Discrete Fourier transforms Optimization Nonlinear distortion 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 << < 14 15 16 17 18 19 20 21 22 23 > >>

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：