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

939 篇 会议
253 篇 期刊文献
3 篇 学位论文

馆藏范围

1,195 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

604 篇 工学
- 425 篇 电气工程
- 185 篇 信息与通信工程
- 172 篇 计算机科学与技术...
- 70 篇 电子科学与技术（可...
- 60 篇 软件工程
- 19 篇 仪器科学与技术
- 18 篇 控制科学与工程
- 17 篇 生物医学工程（可授...
- 8 篇 船舶与海洋工程
- 6 篇 机械工程
- 2 篇 力学（可授工学、理...
- 2 篇 材料科学与工程（可...
- 2 篇 测绘科学与技术
- 1 篇 光学工程
- 1 篇 动力工程及工程热...
- 1 篇 石油与天然气工程
211 篇 理学
- 189 篇 物理学
- 18 篇 系统科学
- 15 篇 生物学
- 4 篇 数学
- 1 篇 化学
- 1 篇 科学技术史(分学科...
64 篇 医学
- 62 篇 临床医学
- 15 篇 特种医学
- 3 篇 基础医学(可授医学...
15 篇 管理学
- 15 篇 管理科学与工程(可...
6 篇 军事学
- 6 篇 军队指挥学
4 篇 艺术学
- 4 篇 音乐与舞蹈学
3 篇 文学
- 2 篇 新闻传播学
1 篇 教育学
- 1 篇 教育学

主题

1,195 篇 audio coding
260 篇 bit rate
189 篇 decoding
174 篇 speech coding
151 篇 quantization
144 篇 transform coding
138 篇 codecs
102 篇 frequency
90 篇 mpeg 4 standard
87 篇 filter bank
80 篇 psychoacoustic m...
77 篇 bandwidth
71 篇 audio compressio...
70 篇 encoding
68 篇 signal processin...
68 篇 discrete cosine ...
56 篇 humans
56 篇 testing
56 篇 streaming media
54 篇 auditory system

机构

19 篇 fraunhofer iis e...
11 篇 institute for in...
9 篇 int audio labs e...
9 篇 dolby sweden ab ...
8 篇 fraunhofer insti...
7 篇 univ calif santa...
7 篇 department of me...
6 篇 philips research...
6 篇 department of el...
6 篇 department of el...
6 篇 fraunhofer insti...
6 篇 department of el...
6 篇 department of el...
6 篇 nokia research c...
5 篇 microsoft corp r...
5 篇 electrical engin...
5 篇 international au...
5 篇 international au...
4 篇 zte corporation
4 篇 huawei technolog...

作者

17 篇 t. moriya
15 篇 ruimin hu
15 篇 tsung-han tsai
15 篇 g. schuller
14 篇 susanto rahardja
12 篇 edler bernd
10 篇 r. geiger
10 篇 te li
9 篇 villemoes lars
9 篇 w. bastiaan klei...
9 篇 r. heusdens
9 篇 s. rahardja
9 篇 j. herre
9 篇 beack seungkwon
9 篇 k. brandenburg
8 篇 x. lin
8 篇 hu ruimin
8 篇 r. vafin
8 篇 a. jin
8 篇 helmrich christi...

语言

1,139 篇 英文
51 篇 其他
4 篇 中文
1 篇 斯洛伐克文

检索条件"主题词=Audio coding"

共 1195 条记录，以下是441-450 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：

Low Delay Filterbanks for Enhanced Low Delay audio coding

Low Delay Filterbanks for Enhanced Low Delay Audio Coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Markus Schnell Ralf Geiger Markus Schmidt Markus Multrus Michael Mellar Jurgen Herre Gerald Schuller Fraunhofer IIS Erlangen Germany Fraunhofer IDMT Ilmenau Germany

Low delay perceptual audio coding has recently gained wide acceptance for high quality communication. While common schemes are based on the well-known Modified Discrete Cosine Transform (MDCT) filterbank, this paper describes novel coding algorithms that, for the first time, make use of dedicated low delay filterbanks, thus achieving improved coding efficiency while maintaining or even reducing the low codec delay. The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) coder currently under development within ISO/MPEG combines a traditional perceptual audio coding scheme with spectral band replication (SBR), both running in a delay-optimized fashion by using low delay filterbanks.

关键词： audio coding Codecs Delay systems Filter bank Design methodology MPEG 4 Standard Delay effects Conferences Acoustic signal processing Acoustic applications

来源：评论

学校读者我要写书评

暂无评论

Lossless audio coding using Burrows-Wheeler Transform and move-to-front coding

Lossless audio coding using Burrows-Wheeler Transform and mo...

引用

International Conference on Computer Engineering and Systems

作者： Hend Ali Elsayed Masoud Alghoniemy Department of Electrical Engineering Alexandria University Alexandria Egypt

ISBN: (纸本)1424413656;9781424413652

In this paper we present a new lossless audio coder using the Burrows-Wheeler transform (BWT). audio signals under consideration, which are assumed to be of integer values, are first permuted using the BWT to get the transform coefficients; and then the resulting coefficients are compressed using a combination of the move-to-front coding (MTF) and an entropy coder. Both Arithmetic and Huffman coding have been used for entropy coding. Simulations show that the proposed coder outperforms other lossless coders, namely, the integer discrete cosine transform (intDCT) and the integer wavelet transform (intWT).

关键词： audio coding Discrete transforms Image reconstruction Discrete wavelet transforms Sorting Entropy coding Streaming media Arithmetic Huffman coding Discrete cosine transforms

来源：评论

学校读者我要写书评

暂无评论

Primary-Ambient Signal Decomposition and Vector-Based Localization for Spatial audio coding and Enhancement

Primary-Ambient Signal Decomposition and Vector-Based Locali...

引用

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

作者： Michael M. Goodwin Jean-Marc Jot Creative Advanced Technology Center Scotts Valley CA USA

Spatial audio coding and enhancement address the growing commercial need to store and distribute multichannel audio and to render content optimally on arbitrary reproduction systems. In this paper, we discuss a spatial analysis-synthesis scheme which applies principal component analysis to an STFT-domain representation of the original audio to separate it into primary and ambient components, which are then respectively analyzed for cues that describe the spatial percept of the audio scene on a per-tile basis; these cues are used by the synthesis to render the audio appropriately on the available playback system. The proposed framework can be tailored for robust spatial audio coding, or it can be applied directly to enhancement scenarios where there are no rate constraints on the intermediate spatial data and audio representation.

关键词： Signal resolution audio coding Principal component analysis Layout Loudspeakers Information analysis Signal synthesis Signal analysis Decoding Robustness

来源：评论

学校读者我要写书评

暂无评论

A Memory-Based Hardware Accelerator for Real-Time MPEG-4 audio coding and Reverberation

A Memory-Based Hardware Accelerator for Real-Time MPEG-4 Aud...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Guo-An Jian Chih-Da Chien Jiun-In Guo Department of Computer Science and Information Engineering National Chung Hsing University Chiayi Taiwan

In this paper we propose a memory-based hardware accelerator for MPEG-4 audio coding and reverberation to achieve both high quality and reality of audio. The proposed design can realize both the computation-intensive component of 256/2048-point IMDCT and the 1024-point FFT-based reverberation through the same hardware engine by adopting a unified IMDCT/FFT/IFFT algorithm, which greatly reduces the hardware cost. The proposed design can achieve both the real-time 5.1 channel audio decoding at the sampling rate of 44.1 KHz and audio reverberation with the hardware cost of 26,633 gates and 4.6K words of local memory for storing transform coefficients and temporary results. The maximum working frequency achieves 220 MHz when implemented by UMC 0.18mum CMOS technology, which can fit the real-time processing requirement of many high quality MPEG-4 audio coding applications

关键词： Hardware MPEG 4 Standard audio coding Reverberation Costs CMOS technology Algorithm design and analysis Engines Decoding Sampling methods

来源：评论

学校读者我要写书评

暂无评论

Extending Fine-Grain Scalable audio coding to Very Low Bitrates using Overcomplete Dictionaries

Extending Fine-Grain Scalable Audio Coding to Very Low Bitra...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Emmanuel Ravelli Gael Richard Laurent Daudet Institut Jean Le Rond d'Alembert LAM Team University Pierre et Marie Curie Paris France TSI Department GET/ENST (Télécom Paris) Paris France Institut Jean Le Rond d'Alembert LAM Team University Pierre et Marie Curie-Paris 6 Paris France

Signal representations in overcomplete dictionaries are considered here as an alternative to the traditional transform representations for fine-grain scalable audio coding. Such representations produce sparser decompositions and thus allow better coding efficiency than transform coding at very low bitrates. Moreover, the decomposition algorithms are intrinsically progressive, and flexible enough to allow an efficient transient modeling. We propose in this paper a fine-grain scalable audio coder which works on a large range of bitrates (2kbs to 128kbs). Objective measures as well as informal subjective evaluation show that this coder outperforms a comparable transform-based coder at very low bitrates.

关键词： audio coding Bit rate Dictionaries MPEG 4 Standard Transform coding Image coding Bandwidth Signal representations Time frequency analysis Conferences

来源：评论

学校读者我要写书评

暂无评论

Application of Unimodular Filterbanks for audio coding

Application of Unimodular Filterbanks for Audio Coding

引用

IEEE Workshop on Signal Processing Systems (SIPS)

作者： Asha Vijayakumar G. Abhilash Department of Electronics and Communication Engineering National Institute of Technology Calicut India

This paper contains the application of Unimodular Filterbanks (UMFBs) based on two new structures for audio coding. Two key properties make UMFBs ideal for audio coding applications. Firstly, they have the lowest possible system delay of M _ 1 samples, where M is the number of channels. This property resolves the problem of pre-echoes. Secondly, the presence of nilpotent matrix in the structure furnishes unequal lengths of analysis and synthesis filters to adapt to perceptual limits; long analysis filters for good frequency selectivity and short synthesis filters for noise reduction. The proposed structures ensure perfect reconstruction (PR) and are of minimum delay in the McMillan sense. The number of free parameters is as low as 2M and the optimal set of parameters can be obtained with unconstrained optimization. The low computational overhead makes it suitable for simple and efficient hardware implementation. We have also carried out a brief study of the quantization noise shaping with the proposed structures.

关键词： audio coding Finite impulse response filter Quantization Table lookup Delay systems Discrete transforms Linear matrix inequalities Frequency synthesizers Noise reduction Image reconstruction

来源：评论

学校读者我要写书评

暂无评论

Sinusoidal modeling of spot microphone signals based on noise transplantation for multichannel audio coding

Sinusoidal modeling of spot microphone signals based on nois...

引用

European Signal Processing Conference (EUSIPCO)

作者： Christos Tzagkarakis Athanasios Mouchtaris Panagiotis Tsakalides Department of Computer Science University of Crete and Institute of Computer Science (FORTH-ICS) Heraklion Crete Greece

This paper focuses on high-fidelity multichannel audio modeling based on an enhanced adaptation of the well-known sinusoidal plus noise model (SNM). Sinusoids cannot be used per se for high-quality audio modeling because they do not represent all the audible information of a recording. The noise part has also to be treated to avoid an artificial sounding resynthesis of the audio signal. Generally, the encoding process needs much higher bitrates for the noise part than the sinusoidal one. Our objective is to model spot microphone signals using the SNM, by taking advantage of the interchannel similarities to achieve low bitrates. We demonstrate that for a given multichannel audio recording, the noise part for each spot microphone signal (microphone signals before the mixing stage) can be obtained from the noise part of one of the signals (reference, which is fully encoded), by transforming the reference noise part using the noise envelope of each of the remaining signals.

关键词： Noise Microphones Computational modeling Bit rate audio coding Harmonic analysis

来源：评论

学校读者我要写书评

暂无评论

Distributed Spatial audio coding in Wireless Hearing Aids

Distributed Spatial Audio Coding in Wireless Hearing Aids

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Olivier Roy Martin Vetterli School of Computer and Communication Sciences Ecole Polytechnique Fédérale de Lausanne Lausanne Switzerland Department of Electrical Engineering and Computer Sciences University of California Berkeley Berkeley CA USA

The information content of binaural signals can be beneficial to many algorithms deployed in current digital hearing aids. However, the exchange of such signals over a wireless communication link requires transmission schemes that must fulfill demanding technical constraints. We present a distributed coding algorithm that builds on psychoacoustic principles in order to achieve this goal with low bitrates, while preserving affordable complexity. The key steps of the proposed algorithm are detailed and the accuracy of the signal exchange mechanism is evaluated in simple simulated acoustic scenarios.

关键词： audio coding Hearing aids Auditory system Signal processing algorithms Bit rate Acoustic devices Distributed computing Wireless communication Psychology Psychoacoustic models

来源：评论

学校读者我要写书评

暂无评论

NOISE-TO-MASK RATIO MINIMIZATION BY WEIGHTED NON-NEGATIVE MATRIX FACTORIZATION

NOISE-TO-MASK RATIO MINIMIZATION BY WEIGHTED NON-NEGATIVE MA...

引用

2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Nikunen, Joonas Virtanen, Tuomas Tampere Univ Technol Dept Signal Proc FI-33720 Tampere Finland

ISBN: (纸本)9781424442966

This paper proposes a novel algorithm for minimizing the perceptual distortion in non-negative matrix factorization (NMF) based audio representation. We formulate the noise-to-mask ratio audio quality criterion in a form where it can be used in NMF and propose an algorithm for optimizing the criterion. We also propose a method for compensating the spreading of the representation error in the synthesis filterbank. The objective perceptual quality produced by the proposed method is found to outperform all the reference methods. We also study the trade-off between the window length and the rank of factorization with a fixed data rate, and find that the best performance is obtained with window lengths between 10 and 30 ms.

关键词： Non-negative matrix factorization Noise-to-mask ratio audio coding Signal representations

来源：评论

学校读者我要写书评

暂无评论

Wide-Band Perceptual audio coding Based on Frequency-Domain Linear Prediction

Wide-Band Perceptual Audio Coding Based on Frequency-Domain ...

引用

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

作者： Petr Motlicek Vijay Ullal Hynek Hermansky Idiap Research Institute Martigny Switzerland International Computer Science Institute Berkeley CA USA école Polytechnique Fédérale de Lausanne Switzerland

In this paper we propose an extension of the very low bit-rate speech coding technique, exploiting predictability of the temporal evolution of spectral envelopes, for wide-band audio coding applications. Temporal envelopes in critically band-sized sub-bands are estimated using frequency domain linear prediction applied on relatively long time segments. The sub-band residual signals, which play an important role in acquiring high quality reconstruction, are processed using a heterodyning-based signal analysis technique. For reconstruction, their optimal parameters are estimated using a closed-loop analysis-by-synthesis technique driven by a perceptual model emulating simultaneous masking properties of the human auditory system. We discuss the advantages of the approach and show some properties on challenging audio recordings. The proposed technique is capable of encoding high quality, variable rate audio signals on bit-rates below 1 bit/sample.

关键词： Wideband audio coding Speech coding Frequency estimation Frequency domain analysis Signal processing Signal analysis Parameter estimation Humans Auditory system

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120页 << < 41 42 43 44 45 46 47 48 49 50 > >>

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：