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121-130 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：

An upper bound on the quality of artificial bandwidth extension of narrowband speech signals

An upper bound on the quality of artificial bandwidth extens...

引用

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

作者： Peter Jax Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

The aim of the artificial bandwidth extension (BWE) of speech signals is to recover wideband speech from bandlimited speech. As the BWE algorithm is supposed to operate without additional side information on the original wideband speech, it has to exploit mutual dependencies between the available and missing frequency bands of the speech signal. In this paper the BWE is examined from an information theoretic perspective. After defining a performance measure, and introducing a few assumptions on a generalized BWE algorithm, a general relationship between mutual information and the maximum achievable estimation performance is formulated, which ensues an upper bound on the performance of BWE algorithms. Finally, some measurements considering a representative BWE scenario are presented.

关键词： Wideband Speech Estimation

来源：评论

学校读者我要写书评

暂无评论

A versatile DSP-system for student-projects on embedded real-time audio signal processing

A versatile DSP-system for student-projects on embedded real...

引用

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

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

In this contribution, a new software environment for an embedded system is presented that is part of a laboratory for teaching students in digital audio signal processing. This environment enables even students with only a limited background in programming embedded technology to achieve remarkable implementations of realtime projects, including runtime user interaction. Hardware and algorithm related programming issues are separated for audio processing and interactive messaging. While the hardware related part is provided along with the software environment, the students can completely focus on algorithmic aspects. A well defined interface builds the bridge between the two programming components. Examples are presented to demonstrate the flexibility and efficiency of the given software platform. The underlying projects were defined, implemented and finally demonstrated by students in an interactive real-time demonstration.

关键词： Signal processing Signal processing algorithms Hardware Embedded software Embedded system Education Digital signal processing Runtime environment Software algorithms Bridges

来源：评论

学校读者我要写书评

暂无评论

A NEW METHOD FOR SNR-ESTIMATION IN IMPULSE RESPONSE MEASUREMENTS

A NEW METHOD FOR SNR-ESTIMATION IN IMPULSE RESPONSE MEASUREM...

引用

European signal processing conference

作者： Aulis Telle Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany

ISBN: (纸本)9781617388767

Methods for measuring the impulse response of a linear transmission system and system identification algorithms in general must be robust against noise in the measured system response. To handle the noise it is of great advantage to know the instantaneous signal-to-noise ratio (SNR), especially in situations with changing noise conditions. In this paper we present a new approach for estimating the SNR during an impulse response measurement by means of the so-called sliding window correlation (SWiC) as introduced in this paper. The performance of the proposed method is evaluated by means of simulation results.

关键词： IMPULSE RESPONSE new methods system response Transmission systems measuring method

来源：评论

学校读者我要写书评

暂无评论

Analysis-by-synthesis speech coding with extended approximation interval

Analysis-by-synthesis speech coding with extended approximat...

引用

IEEE Workshop on Speech Coding For Telecommunications

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

In analysis-by-ryndzesis coders the problem of approximating the Original signal by the synthesized signal is solved over a limited time interval only. In this contribution a systematic investigation of possible impro... 详细信息

关键词： Speech analysis Speech coding Finite impulse response filter Signal synthesis Quantization Signal processing IIR filters Codecs communication systems data processing

来源：评论

学校读者我要写书评

暂无评论

Robust speech decoding: can error concealment be better than error correction?

Robust speech decoding: can error concealment be better than...

引用

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

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

Digital speech transmission systems use source coding to reduce the bit rate and channel coding to correct transmission errors. Furthermore, in periods of a very poor channel quality error concealment of residual bit errors becomes necessary as channel decoding fails. However, if the channel is clear, channel coding would not be required at all and the speech quality could be improved by allowing a higher bit rate for source encoding. Usually a compromise is taken between speech quality in case of a clear channel and error robustness in case of poor channel quality. This paper addresses the problem of a joint optimization of error concealment and source/channel coding. Under the premise of a minimum mean square error criterion for signal reconstruction it turns out that error concealment instead of error correction may be the best choice if source coding leaves sufficient residual parameter correlations by less bit rate reduction.

关键词： Robustness Decoding Bit rate Channel coding Speech coding Source coding Error correction Mean square error methods Signal reconstruction Error correction codes

来源：评论

学校读者我要写书评

暂无评论

A novel psychoacoustically motivated audio enhancement algorithm preserving background noise characteristics

A novel psychoacoustically motivated audio enhancement algor...

引用

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

作者： S. Gustafsson P. Jax P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In this paper we propose an algorithm for reduction of noise in audio signals. In contrast to several previous approaches we do not try to achieve a complete removal of the noise, but instead our goal is to preserve a pre-defined amount of the original noise in the processed signal. This is accomplished by exploiting the masking properties of the human auditory system. The speech and noise distortions are considered separately. The spectral weighting rule, adapted by utilizing only estimates of the masking threshold and the noise power spectral density, has been designed to guarantee complete masking of distortions of the residual noise. Simulation results confirm that no audible artifacts are left in the processed signal, while speech distortions are comparable to those caused by conventional noise reduction techniques.

关键词： Psychology Acoustic noise Noise reduction Signal processing Speech enhancement Humans Auditory system Masking threshold Psychoacoustic models Speech processing

来源：评论

学校读者我要写书评

暂无评论

Design of IIR QMF banks with near-perfect reconstruction and low complexity

Design of IIR QMF banks with near-perfect reconstruction and...

引用

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

作者： Heinrich W. Lollmann Peter Vary Institute of Communication Systems and Data Processing (IND) RWTH Aachen University Aachen Germany

A novel design for a two-channel IIR quadrature-mirror filter (QMF) bank with near-perfect reconstruction (NPR) is presented. The analysis filter-bank is given by an efficient polyphase network (PPN) implementation based on allpass filters. The arising phase distortions are almost compensated by stable allpass filters, designed via analytical closed-form expressions. In a first design, the remaining aliasing, amplitude and phase distortions become arbitrarily small in dependence of the tolerable system delay and algorithmic complexity, respectively. In a second design, aliasing and amplitude distortions are completely canceled and phase distortions are minimized at the expense of an additional signal delay. The proposed QMF banks have a lower algorithmic complexity than comparable designs.

关键词： Finite impulse response filter IIR filters Phase distortion Signal synthesis Filter bank Computational complexity Transfer functions Closed-form solution Algorithm design and analysis Delay systems

来源：评论

学校读者我要写书评

暂无评论

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

Noise reduction by joint maximum a posteriori spectral amplitude and phase estimation with super-Gaussian speech modelling

Noise reduction by joint maximum a posteriori spectral ampli...

引用

European Signal processing Conference (EUSIPCO)

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

For acoustical background noise reduction a computationally efficient joint MAP estimator with a super-Gaussian speech model is presented. Compared to a recently introduced MAP estimator the new joint MAP estimator allows an optimal adjustment of the underlying statistical model to the real PDF of the speech spectral amplitude. The computationally efficient estimator outperforms the Ephraim-Malah estimator and the recently proposed MAP estimator in a single microphone noise reduction framework due to the more accurate statistical model.

关键词： Noise Abstracts Rician channels

来源：评论

学校读者我要写书评

暂无评论

A probabilistic framework for optimum speech extrapolation in digital mobile radio

A probabilistic framework for optimum speech extrapolation i...

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 digital mobile radio systems the speech quality can be degraded severely if the channel decoder produces residual bit errors, e.g., due to heavy burst errors on the channel. A novel combined speech extrapolation and error detection algorithm is presented which can improve the speech significantly in the case of residual bit errors. This algorithm, which is part of the speech decoding process, uses a posteriori probabilities of speech parameters. With the extracted a posteriori probability, optimum estimators adapted to human perception can be applied and soft decision information can be exploited fully. In terms of perceptual performance, the MS (mean-square) estimator is superior to the MAP (maximum a posteriori) estimator. The method was tested under realistic conditions using an 8-kbit/s CELP (code excited linear prediction) codec. A significant improvement of subjective speech quality can be achieved.< >

关键词： Extrapolation Decoding Speech processing Land mobile radio Degradation Detection algorithms data mining Humans Testing Codecs

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 << < 9 10 11 12 13 14 15 16 17 18 > >>

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：