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1141-1150 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：

Digital audio compression technology and AVS audio standard research

Digital audio compression technology and AVS audio standard ...

引用

IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： RuiMin Hu Yong Zhang Haojun Ai National Multimedia Software Engineering Technology Research Center Wuhan University of China Wuhan China

This paper gives a brief overview of the current research of digital audio coding and decoding standard, and introduces the newly established AVS audio standard. It also draws the roadmap of AVS/spl ***/M audio standa... 详细信息

关键词： audio compression audio coding Transform coding Standards development Bit rate Multimedia systems Software standards Humans DVD Bandwidth

来源：评论

学校读者我要写书评

暂无评论

Scalable audio coder based on quantizer units of MDCT coefficients

Scalable audio coder based on quantizer units of MDCT coeffi...

引用

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

作者： A. Jin T. Moriya T. Norimatsu M. Tsushima T. Ishikawa NTT Human Interface Laboratories Musashino Tokyo Japan Matsushita Elecrric Indusrrial Company Limited Kadoma Osaka Japan

A scalable codec has been constructed by using transform coding and the basic modules for scalable encoder and decoder. It allows users to choose a variety of scalable configurations in the frequency domain. The basic module is a quantizer that can quantize MDCT (modified DCT) coefficients transformed from a variety of frequency regions. This module mainly works at bit rates of more than 8 kbit/s. We can also change the target frequency regions of the basic module's input-output signals in each transform frame; i.e., we can change the scalable structure according to the nature of the input signals. In the scalable codec described here, the input-output signals are monaural and the sampling frequency is 24 kHz. The total bit rate of this scalable codec is more than 8 kbit/s. Subjective quality evaluation tests, mainly for musical sound sources, showed that it's sound quality is better than that of an MPEG-2 layer 3 codec at 8, 16, and 24 kbit/s when our scalable codec is constructed of 8-kbit/s basic modules. In combination with AAC (advanced audio coding), our scalable codec will be chosen as an international standard in ISO/IEC-MPEG-4/audio.

关键词： Codecs Bit rate ISO standards Transform coding Decoding Frequency domain analysis Discrete cosine transforms Sampling methods Acoustic testing audio coding

来源：评论

学校读者我要写书评

暂无评论

Multiband amplitude modulated sinusoidal audio modeling

Multiband amplitude modulated sinusoidal audio modeling

引用

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

作者： M.G. Christensen S. van de Par S.H. Jensen S.V. Andersen Dept. of Communication Technology Aalborg University Denmark Philips Research Laboratories Eindhoven The Netherlands

In this paper, we investigate the importance of taking frequency-dependent temporal phenomena into account in audio coding. We do this in the context of sinusoidal modeling of audio signals by applying amplitude modulation to the sinusoidal components. Traditionally, audio coders use a fixed time-segmentation for all frequencies despite the fact that it is well-known that the time-frequency resolution of the human auditory system is not constant. The well-known window switching is an example of this. We compare multiband amplitude modulated sinusoidal models to a singleband model using different audio excerpts. Based on both comparative listening tests and a psychoacoustical distortion measure it is concluded that an improvement is generally gained using multiband amplitude modulation, although specific single sources are well-modeled using a singleband model.

关键词： Amplitude modulation Psychoacoustic models Distortion measurement audio coding Context modeling Time frequency analysis Signal resolution Humans Auditory system Testing

来源：评论

学校读者我要写书评

暂无评论

audio representations on overcomplete sets

Audio representations on overcomplete sets

引用

International Conference on Digital Signal Processing (DSP)

作者： L. Daudet M. Sandler B. Torresani Department of Electronic Engineering Queen''s Mary University of London London UK LATP-CMI Université de Provence Marseilles France

We present in this paper the advantages of using overcomplete sets for the representation and coding of audio signals. More specifically, we have investigated hybrid signal models of the type {tonal+transients+stochastic residual} where these features can be simultaneously present at all times. The extraction of the two "deterministic" components (tonal and transient parts) is made in such a way that each of them can be represented in a compact way in appropriate orthogonal basis (MDCT and dyadic wavelets, respectively). The use of structured representations significantly reduces the potentially high cost of encoding of significance maps. Preliminary results show that this approach is well-adapted for transform coding of very high-quality sounds.

关键词： Costs audio coding Stochastic processes Encoding Transform coding Transient analysis Stress Acoustic noise Contracts Instruments

来源：评论

学校读者我要写书评

暂无评论

ROBUST CRITICAL DATA RECOVERY FOR MPEG-4 AAC ENCODED BITSTREAMS

ROBUST CRITICAL DATA RECOVERY FOR MPEG-4 AAC ENCODED BITSTRE...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Ruijing Hu Xucen Huang Michel Kieffer Olivier Derrien Pierre Duhamel L2S CNRS-SUPELEC-Univ Paris-Sud 3 rue Joliot-Curie 91192 Gif-sur-Yvette France LMA CNRS Marseille France

ISBN: (纸本)9781424442959;9781424442966

This paper presents a bandwidth-efficient method for improved decoding of critical data generated by the MPEG-4 AAC audio coder when encoded bitstreams are transmitted over noisy channels. The critical data of each encoded frame is estimated using the redundancy due to the correlation between successive frame headers and using an optional CRC as an error-correcting code. Simulation results for an AWGN channel show a substantial link budget improvement (more than 5 dB) compared to a classical hard-decoding method. Once critical data are efficiently estimated, the work of previously proposed joint source-channel decoding techniques to recover the remaining parts of the MPEG4 AAC frames is significantly facilitated.

关键词： audio coding Cyclic codes Decoding MAP estimation Mobile communication Robustness audio coding cyclic codes DEcoding mobile communication MPEG 4 Standard automatic amplitude control AWGN channels Robustness CRITICAL

来源：评论

学校读者我要写书评

暂无评论

Improved Algorithms for Implementation of MPEG2 AAC Decoder on FPGA

Improved Algorithms for Implementation of MPEG2 AAC Decoder ...

引用

IEEE Region 10 International Conference TENCON

作者： Rajath R. Shenoy Sudhir S. Naik D.S. Sumam Deptartment of Electronics & Communication Engineering National Institute of Technology Karnataka Surathkal Mangalore INDIA Dept. of Electron. & Commun. Eng. Nat. Inst. of Technol. Karnataka Mangalore

This paper discusses methods to implement the IMDCT filter bank, noiseless decoder, inverse quantiser and scale factor application modules of MPEG-2 advanced audio coding decoder more efficiently when implemented on F... 详细信息

关键词： Decoding Field programmable gate arrays Filter bank Gain control Streaming media Digital signal processing audio coding Transform coding Digital audio players Noise shaping

来源：评论

学校读者我要写书评

暂无评论

A new perceptual entropy-based method to achieve a signal adapted wavelet tree in a low bit rate perceptual audio coder

A new perceptual entropy-based method to achieve a signal ad...

引用

European Signal Processing Conference (EUSIPCO)

作者： N. Ruiz Reyes M. Rosa Zurera F. López Ferreras D. Martínez Muñoz Departamento de Electrónica Universidad de Jaén Escuela Universitaria Politénica. 23700 Linares - Jaén (Spain) Dpto. de Teoría de la Señal y Comunicaciones Universidad de Alcalá Escuela Politécnica 28871 Alcalá de Henares - Madrid (Spain)

This paper outlines a wavelet based audio coding scheme that uses a perceptual entropy criterion to achieve a signal adapted wavelet tree. We are interested in decompositions adapted to the nature of audio signals, but attending to the characteristics of the human hearing system. Our audio coder is compared with two other wavelet based coding schemes: the first one implements a fixed wavelet decomposition which closely mimics the critical band division at the inner ear, while the second one uses an adaptive wavelet tree controlled by a classical entropy criterion, such as Shannon entropy. The results confirm that perceptual entropy-based decomposition is the way you must take if you are looking for maximum compression rates and transparent coding. To achieve a further bit rate reduction ensuring transparent coding, our audio coder performs a filter order optimization, which is possible by using a novel algorithm to translate the psycho-acoustic information from the frequency to the wavelet domain. Experimental results indicate that the proposed coder ensures transparent coding of monophonie CD quality audio signals at bit rates lower than 64 kb/s (1.45 bit/sample) for most audio signal. Subjective listening tests tell us that our coding method behaves as well as ISO/MPEG 1 — layer III and much better than MPEG 2-layer II at that bit rate.

关键词： Bit rate Wavelet packets Channel coding Cost function audio coding Transform coding

来源：评论

学校读者我要写书评

暂无评论

An overview of MDCT for Time Domain Aliasing Cancellation

An overview of MDCT for Time Domain Aliasing Cancellation

引用

International Conference on Communication and Network Technologies (ICCNT)

作者： V. Arun Raj M. Davidson Kamala Dhas D. Gnanadurai Mepco Schlenk Engineering College Sivakasi India Department of ECE Mepco Schlenk Engineering College Sivakasi India J.P. College of Engineering Ayikudi India

Modified Discrete Cosine Transform (MDCT) is a upcoming special transform implemented in areas of audio signal processing and compression. As, the name implies it is the modified form of DCT (Discrete Cosine Transform) that allows overlapping of segments (say 50%) and thereby it helps in avoiding artifacts. MDCT in general is a lapped transform that rectifies the problem of TDAC (Time Domain Aliasing Cancellation) and hence widely used in audio codec (.mp3, .wav) applications. The scope of this paper is to use MDCT in a system that overcomes the problem of TDAC and implementation of different windows those are capable with this transform. Finally, we calculate the MSE (Mean Squared Error) for the system with different inputs stating that perfect audio reconstruction is possible.

关键词： Time-domain analysis Discrete cosine transforms Speech Speech processing audio coding

来源：评论

学校读者我要写书评

暂无评论

A low bit-rate audio coder based on modified sinusoidal model

A low bit-rate audio coder based on modified sinusoidal mode...

引用

International Conference on Communications, Circuits and Systems (ICCCAS)

作者： Song Shaopeng Yin Junxun Yu Yongcong A.-M. Raed Department of Electronic Engineering South China University of Technology Guangzhou China

A very low bit-rate audio coding algorithm suitable for wireless mobile communications is presented. The codec is based on a sinusoidal model, with modifications to reduce bit-rate and improve reconstructed audio quality. The improvements include signal adaptive analysis length, psychoacoustic model used in both matching pursuits algorithm and parameter quantization, and phaseless frequency domain audio reconstruction. At general bit-rate of 0.5 bit per sample, AM broadcasting audio quality is achieved.

关键词： Psychoacoustic models Matching pursuit algorithms audio coding Wireless communication Mobile communication Codecs Signal analysis Algorithm design and analysis Pursuit algorithms Quantization

来源：评论

学校读者我要写书评

暂无评论

Local bases yielding arbitrary tilings of the time-frequency plane

Local bases yielding arbitrary tilings of the time-frequency...

引用

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

作者： R. Bernardini J. Kovacevic Università di Padova Padova Italy AT and T Bell Laboratories Inc. Murray Hill NJ USA

We show how to obtain arbitrary tilings of the time-frequency plane using local orthogonal bases. These bases were constructed as a generalization of the cosine-modulated filter banks in discrete time, and local trigonometric bases in continuous time. Due to the fact that they use a single prototype window, these bases also lead to time-varying tilings. Moreover, they have a fast implementation algorithm, and allow for multidimensional irreducible basis functions. We show examples of design, in particular, that of a critical-band system for use in audio coding.

关键词： Time frequency analysis Bismuth Filter bank Prototypes audio coding Signal analysis Multidimensional systems Signal design Bars Wavelet analysis

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 << < 111 112 113 114 115 116 117 118 119 120 > >>

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：