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131-140 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：

Very Low Complexity CELP Speech Coding with Adaptive Phase Codebooks

Very Low Complexity CELP Speech Coding with Adaptive Phase C...

引用

IEEE Workshop on Speech Coding For Telecommunications

作者： C.G. Gerlach Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

来源：评论

学校读者我要写书评

暂无评论

CELP speech coding with almost no codebook search

CELP speech coding with almost no codebook search

引用

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

作者： C.G. Gerlach Institute for Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In analysis-by-synthesis speech coders the computational complexity of the search for an optimum innovation is still high although transformations were proposed to decrease the complexity. This limits practical codebook sizes and vector dimensions (block lengths). In this contribution two new structured frequency domain codebooks are proposed. The first one is a pulse shaped codebook with a reversed search order for the gain and the shape; the second one is a unity magnitude codebook with structured phase. The corresponding algorithms which are based on new insights, result in a drastically reduced search in the transformed domain. The computational complexity increases only proportional to the bit rate and not to the codebook size.< >

关键词： Speech coding Bit rate Nonlinear filters Computational complexity Speech synthesis Filtering Speech analysis Technological innovation Pulse shaping methods Wideband

来源：评论

学校读者我要写书评

暂无评论

IIR QMF-bank design for speech and audio subband coding

IIR QMF-bank design for speech and audio subband coding

引用

IEEE Workshop on Applications of Signal processing to Audio and Acoustics

作者： Heinrich W. Löllmann Matthias Hildenbrand Bernd Geiser Peter Vary Institute of Communication Systems and Data Processing (IND) RWTH Aachen University Aachen Germany

A new speech and audio codec has been submitted recently to ITU-T by a consortium of Huawei and ETRI as candidate proposal for the super-wideband and stereo extensions of ITU-T Rec. G.729.1 and G.718. This hierarchical codec with bit rates from 8-64 kbit/s relies on a subband splitting by means of a quadrature-mirror filter-bank (QMF-bank). For this, an allpass-based QMF-bank is used whose design and implementation is presented in this contribution. This IIR filter-bank allows to achieve a significantly lower signal delay in comparison to the traditional FIR QMF-bank solution without a compromise for the speech and audio quality.

关键词： Speech coding Finite impulse response filter IIR filters Delay Proposals Speech codecs Bit rate Decoding Audio coding Speech processing

来源：评论

学校读者我要写书评

暂无评论

Robust speech decoding: a universal approach to bit error concealment

Robust speech decoding: a universal approach to bit error co...

引用

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

作者： T. Fingscheidt P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In digital mobile communication systems there is the need for reducing the subjective effects of residual bit errors which have not been eliminated by channel decoding by the use of error concealment techniques. Due to the fact that most standards do not specify these algorithms bit exactly, there is room for new solutions to improve the speech quality. This article develops a new approach for optimum estimation of the speech codec parameters. It can be applied to any speech codec standard if bit reliability information is provided by the demodulator (e.g. DECT), or by the channel decoder (e.g. soft-output Viterbi algorithm-SOVA in GSM). The proposed method includes an inherent muting mechanism leading to a graceful degradation of speech quality in case of adverse transmission conditions. Particularly the additional exploitation of the residual source redundancy, i.e. some a priori knowledge about the codec parameters gives a significant enhancement of the output speech quality. In the case of an error free channel, bit exactness as required by the standards can be preserved.

关键词： Robustness Decoding Speech codecs Mobile communication Code standards Demodulation Viterbi algorithm GSM Degradation Redundancy

来源：评论

学校读者我要写书评

暂无评论

Uplink performance of a new macro-diversity cellular mobile radio architecture

Uplink performance of a new macro-diversity cellular mobile ...

引用

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

作者： W. Papen Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

来源：评论

学校读者我要写书评

暂无评论

Approximation of optimal step size control for acoustic echo cancellation

Approximation of optimal step size control for acoustic echo...

引用

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

作者： C. Antweiler J. Grunwald H. Quack Institute for Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

One of the most widely used gradient-based adaptation algorithms is the so called normalized least mean square (NLMS) algorithm. The rate of convergence, misadjustment and noise insensitivity of the NLMS-type algorithm depend on the proper choice of the step size parameter, which controls the weighting applied to each coefficient update. Different step size methods have been proposed to improve the convergence of NLMS-type filters, while preserving the steady-state performance. The step size methods considered here use either a step size parameter which varies with time or a separate, tap-individual step size for each filter tap. The derivation of the respective step size methods is based on different optimization criteria. In this paper a step size parameter is proposed satisfying a combined optimization criterion leading to a time variant and individual step size parameter. The realization aspects of the new concept are discussed for an acoustic echo control application as an example.

关键词： Size control Echo cancellers Convergence Steady-state communication system control Adaptive filters Least squares approximation data processing Acoustic noise Noise cancellation

来源：评论

学校读者我要写书评

暂无评论

16 kbit/s wideband speech coding based on unequal subbands

16 kbit/s wideband speech coding based on unequal subbands

引用

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

作者： J.W. Paulus J. Schnitzler Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

We propose a split-band encoding scheme for 16 kbit/s wideband speech coding (50-7000 Hz), using 2 unequal subbands from 0-6 kHz and from 6-7 kHz. This approach was motivated by an experimental evaluation of the signal bandwidth of speech frames. The higher subband is simply represented by white noise with adjustment of the short term energy. For the lower subband code-excited linear prediction (CELP) is used. The analysis filter bank, which performs the unequal band splitting combined with critical subsampling of the sub-bands, is described. A bit error concealment technique and the bit allocation is also presented. By informal listening tests the speech quality was rated higher than the speech quality of the CCITT G.722 wideband codec operating at 48 kbit/s.

关键词： Wideband Speech coding Speech codecs Encoding Bandwidth Speech analysis White noise Performance analysis Filter bank Bit rate

来源：评论

学校读者我要写书评

暂无评论

Beamformer for Driving Binaural Speech Enhancement

Beamformer for Driving Binaural Speech Enhancement

引用

IWAENC 2012;International Workshop on Acoustic Signal Enhancement

作者： Heinrich W. Loellmann Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany

A beamformer for binaural speech enhancement systems in digital hearing aids is proposed. Its single modules for the estimation of the time-difference-of-arrival (TDOA) and time-alignment operate in the frequency-domain and have a low computational complexity. The TDOA estimation is performed efficiently by a generalized cross-correlation with phase transform weighting. The estimation accuracy for filter-banks with a limited number of subbands, which are needed for hearing aids to meet tight delay constraints, is improved by a histogram-based TDOA estimation. The subsequent time-alignment is accomplished by a simple multiplication with spectral phase factors. A primary application of the proposed system are binaural cue preserving speech enhancement systems based on spectral weighting. The proposed beamformer can be used as delay-and-sum and/or delay-and-subtract beamformer to provide subband signals from which the power spectral densities of interfering sources can be estimated to drive the spectral weight calculation.

关键词： Estimation Speech Speech enhancement Delay Complexity theory Discrete Fourier transforms Algorithm design and analysis

来源：评论

学校读者我要写书评

暂无评论

New insights into the statistical signal model and the performance bounds of acoustic echo control

New insights into the statistical signal model and the perfo...

引用

European Signal processing Conference (EUSIPCO)

作者： Gerald Enzner Peter Vary Institute of Communication Systems and Data Processing (IND) Aachen University (RWTH) Aachen Germany

The contribution of this paper is two-fold. At first, we introduce a modification of the linear statistical signal model in acoustic echo control. In contrast to the traditional approach, the acoustic echo path is characterized as a random process with statistical mean and covariance, while the echo path input is modeled as a deterministic signal. Based on the modified signal model, we then derive the linear MMSE estimator for the near-end speech components in the microphone signal. The result can be seen as a generalized Wiener filter that consists of an acoustic echo canceler and a post-filter for residual echo suppression. The presented theory entails several fundamental advantages: a) the new signal model better matches the practical applications of acoustic echo control, b) it proves the principal coexistence of echo canceler and postfilter in hands-free communication systems, c) the generalized Wiener solution simplifies the realization of acoustic echo controllers, and d) we obtain a better insight into the performance bounds of acoustic echo control.

关键词： Speech Mathematical model Attenuation communication systems Random processes Echo cancellers

来源：评论

学校读者我要写书评

暂无评论

Advanced signal processing in speech communication

Advanced signal processing in speech communication

引用

European Signal processing Conference (EUSIPCO)

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

In digital mobile communications several measures are taken beyond speech coding to enhance the perceived quality in the presence of acoustic background noise and transmission errors. In premium mobile phones meanwhile advanced algorithms for noise suppression, error concealment, and finally artificial bandwidth extension come to practical application. It will be shown in this contribution that theses three different concepts of speech enhancement are actually based on the same common principle of conditional estimation, taking statistical a priori knowledge into account. Recent developments in these three areas are presented.

关键词： Degradation Abstracts GSM Filtering Speech Indexes

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 << < 10 11 12 13 14 15 16 17 18 19 > >>

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：