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
1 篇 学位论文

馆藏范围

1,193 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

1,193 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

1,137 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 1193 条记录，以下是321-330 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：

Binaural reproduction for Directional audio coding

Binaural reproduction for Directional Audio Coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Mikko-Ville Laitinen Ville Pulkki Department Signal Processing and Acoustics Helsinki University of Technology Espoo Finland

Directional audio coding (DirAC) is a recently proposed method for spatial sound reproduction. So far it has been used only in loud-speaker reproduction, and a method for headphone reproduction is presented in this article. In principle, the method uses virtual loud-speakers simulated with head related transfer functions (HRTFs), and head tracking. The method was evaluated subjectively, and the results are presented. The results show that a plausible spatial impression can be reproduced using the binaural realization of DirAC.

关键词： audio coding Loudspeakers Microphones Frequency Signal analysis Acoustic signal processing Acoustic applications Signal synthesis Conferences Headphones

来源：评论

学校读者我要写书评

暂无评论

Low delay moving-horizon multiple-description audio coding forwireless hearing aids

Low delay moving-horizon multiple-description audio coding f...

引用

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

作者： Jan Ostergaard Daniel E. Quevedo Jesper Jensen Department of Electronic Systems University of Aalborg Aalborg Denmark School of Electrical Engineering and Computer Science University of Newcastle Australia Oticon Copenhagen Denmark

ISBN: (纸本)9781424423538

In this work, we construct a novel scheme for efficient perceptual coding of audio for robust communication between encoders and wireless hearing aids. To limit the physical size of the hearing aids and to reduce power consumption and thereby increase the lifetime expectancy of the batteries, the hearing aids are constrained to be of low complexity. We therefore provide an asymmetric strategy where most of the computational load is placed at the encoding side. We make use of multiple-description coding. This combats possible erasures on the wireless link between the encoder and the hearing aids without introducing significant delay. Furthermore, we employ psychoacoustically optimized noise-shaping quantizers based on the moving-horizon principle, which exploits a finite prediction horizon.

关键词： Delay audio coding Hearing aids Psychoacoustic models Decoding Robustness Quantization Filters Batteries Noise shaping

来源：评论

学校读者我要写书评

暂无评论

SPATIAL audio coding BY SQUEEZING: ANALYSIS AND APPLICATION TO COMPRESSING MULTIPLE SOUNDFIELDS

SPATIAL AUDIO CODING BY SQUEEZING: ANALYSIS AND APPLICATION ...

引用

European signal processing conference

作者： Bin Cheng Christian Ritz Ian Burnett Whisper Laboratories School of Electrical Computer and Telecommunications Engineering University of Wollongong Wollongong NSW Australia 2500 School of Electrical and Computer Engineering Royal Melbourne Institute of Technology Melbourne VIC Australia 3000

ISBN: (纸本)9781617388767

Spatially Squeezed Surround audio coding (S^AC) proposed by the authors provides efficient compression of multi-channel surround audio. Compression is achieved by exploiting human sound localisation blur to save the surround soundfleld information in a squeezed stereo soundfleld. In this paper, the localisation loss during the S^AC analysis/synthesis is evaluated and the minimum size of the S3AC squeezed soundfleld is derived in a frequency dependent form. Results from perceptual listening tests show that, compared with standard squeezing from a 360° surround soundfleld to a 60°stereo soundfleld, a more intensive squeezing method, such as from 360°-to-5°, does not introduce audible localisation distortion. This leads to a further application of^AC for compressing more than one surround soundfleld into a single stereo downmix for applications such as spatialised teleconferencing. This application is also described and perceptually evaluate.

关键词： minimum dimension audio audio compression audio coding COPPER ALLOY NO 360 MULTICHANNEL Sound Localization Teleconference

来源：评论

学校读者我要写书评

暂无评论

A perceptually enhanced Scalable-to-Lossless audio coding scheme and a trellis-based approach for its optimization

A perceptually enhanced Scalable-to-Lossless audio coding sc...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Emmanuel Ravelli Vinay Melkote Kenneth Rose Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

Scalable-to-lossless (SLS) audio compression, as standardized by MPEG, provides a lossy base layer compatible with the advanced audio coding (AAC) format, ensuring state-of-the-art quality in the base layer, and additional fine grained enhancements that eventually provide a lossless compressed version of the signal. While SLS offers highly efficient lossless compression, the perceptual quality of its intermediate lossy layers has been observed to be suboptimal. This paper proposes a modified SLS audio coding scheme that provides enhanced perceptual quality at an intermediate bit-rate, at the expense of an additional parameter per frequency band as side-information. This scheme when coupled with a trellis-based optimization algorithm is demonstrated to outperform, in terms of quality at the intermediate bit-rate, both standard SLS and a recent perceptually enhanced variant, with minimal degradation in lossless coding performance.

关键词： audio coding Laser sintering Rate-distortion Encoding Performance loss MPEG 4 Standard Discrete transforms Discrete cosine transforms Conferences Application software

来源：评论

学校读者我要写书评

暂无评论

Spatial filtering using directional audio coding parameters

Spatial filtering using directional audio coding parameters

引用

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

作者： Markus Kallinger Giovanni Del Galdo Fabian Kuech Dirk Mahne Richard Schultz-Amling Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

ISBN: (纸本)9781424423538

Directional audio coding (DirAC) is a recent method for spatial audio processing, based on a perceptually motivated representation of spatial sound. Due to its efficiency, DirAC has already been proposed for spatial audio teleconferencing scenarios. Modern hands-free communication systems usually include beamforming techniques to improve speech intelligibility by suppressing diffuse background noise and interfering sources. In this paper, we propose a novel spatial filtering method which can be integrated into the DirAC spatial codec. It uses a spectral weighting of the recorded audio signal, where the design of the corresponding spatial filter transfer function is based on the DirAC parameters, i. e., direction-of-arrival and diffuseness of the sound field. Simulation results show that compared to a standard beamformer the novel technique offers significantly higher interference attenuation, while introducing similar distortion of the desired signal.

关键词： Filtering audio coding Teleconferencing Array signal processing Speech enhancement Background noise Codecs Signal design Spatial filters Transfer functions

来源：评论

学校读者我要写书评

暂无评论

Efficient merging of multiple audio streams for spatial sound reproduction in Directional audio coding

Efficient merging of multiple audio streams for spatial soun...

引用

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

作者： Giovanni Del Galdo Fabian Kuech Markus Kallinger Richard Schultz-Amling Audio Department Fraunhofer IIS Erlangen Germany

ISBN: (纸本)9781424423538

Directional audio coding (DirAC) is an efficient technique to capture and reproduce spatial sound. The analysis step outputs a mono DirAC stream, comprising an omnidirectional microphone pressure signal and side information, i.e., direction of arrival and diffuseness of the sound field expressed in time-frequency domain. This contribution proposes a method to merge two or more mono DirAC streams for a joint playback at the reproduction side. This problem arises in applications such as immersive spatial audio teleconferencing. With respect to a trivial direct merging, the proposed method is more efficient as it does not require the synthesis step. From this follows the benefit that the loudspeaker setup at the reproduction side does not have to be known in advance. Simulations and informal listening tests confirm the absence of any artifacts and that the proposed method is practically indistinguishable from the ideal merging.

关键词： Merging Streaming media audio coding MONOS devices Information analysis Signal analysis Microphones Time frequency analysis Teleconferencing Loudspeakers

来源：评论

学校读者我要写书评

暂无评论

G.722 ANNEX D AND G.711.1 ANNEX F - NEW ITU-T STEREO CODECS

G.722 ANNEX D AND G.711.1 ANNEX F - NEW ITU-T STEREO CODECS

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： David Virette Yue Lang Lei Miao Wenhai Wu Balazs Kovesi Claude Lamblin Stephane Ragot Huawei Technologies China France Telecom Orange France

ISBN: (纸本)9781479903573

This paper presents the two new ITU-T Recommendations G.722 Annex D and G.711.1 Annex F, which are stereo extensions of the wideband codecs ITU-T G.722 and G.711.1 and their superwideband extensions (G.722 Annex B and G.711.1 Annex D). An embedded scalable structure is used to add stereo extension layers on top of the wideband or superwideband core coding. Wideband stereo modes are supported at the bit rates of 64/80 and 96/128 kbit/s for G.722 and G.711.1 (respectively), while superwideband stereo modes are supported at 80/96/112/128 and 112/128/144/160 kbit/s. The parametric stereo coding model is based on a frequency domain downmix, wideband interchannel differences estimation, quantization and synthesis, low complexity coherence analysis and synthesis, stereo transient detection and stereo post-processing. An overview of formal ITU-T characterization listening tests illustrates the performance of these codecs.

关键词： speech coding audio coding parametric stereo coding G.722 Annex D G.711.1 Annex F

来源：评论

学校读者我要写书评

暂无评论

Multi-channel audio Compression Method Based on ITU-T G.719 Codec

Multi-channel Audio Compression Method Based on ITU-T G.719 ...

引用

International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIHMSP)

作者： Wenjin Jiang Jing Wang Yi Zhao Baoguang Liu Xuan Ji School of Information and Electronics Beijing Institute of Technology Beijing China Institute of Acoustics Chinese Academy of Sciences Beijing China Beijing Institute of Technology Beijing Beijing CN

Through exploiting the human perception of spatial sound, a new approach for compression coding of multi-channel audio signal based on ITU-T G.719 codec is put forward in this paper. Multi-channel input signals are converted to a down-mixed signal plus spatial perceptual parameters by use of down-mix and up-mix step-by-step techniques in frequency domain. The algorithm can significantly reduce the coding rate under the premise of an acceptable sound quality in combination with the G.719 audio codec. The paper presents the implementation of the algorithm and describes in detail the calculation and features of the selected spatial parameters. Finally some experiments are done to evaluate the algorithm from the perspective of the compression ratio, the reconstructed sound quality, and the algorithm complexity.

关键词： Codecs Complexity theory Quantization (signal) Decoding audio coding audio compression

来源：评论

学校读者我要写书评

暂无评论

Technologies for speech and audio coding

Technologies for speech and audio coding

引用

IEEE International Symposium on Consumer Electronics (ISCE)

作者： Takehiro Moriya NTT Communication Science Laboratories Nippon Telegraph and Telephone Corporation Atsugi Japan

This paper reviews some important technologies for speech and audio coding, contrasting time domain and frequency domain processing. These technologies have contributed to the standardization activities in ITU-T and ISO/IEC MPEG, respectively. It is interesting that the standardization targets for speech and audio have started to converge recently. The MPEG USAC (Unified Speech and audio coding) scheme, which is being actively developed, may make use of most of the important technologies for speech and audio that have been developed so far.

关键词： Speech coding audio coding Frequency domain analysis Standardization Speech processing Narrowband Bit rate Delay Bandwidth Linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

Distributed audio coding with Efficient Source Correlation Extraction

Distributed Audio Coding with Efficient Source Correlation E...

引用

IEEE Digital Signal Processing Workshop

作者： Sandeep Matta Charles D. Creusere Klipsch School of Electrical & Computer Engineering New Mexico State University Las Cruces NM USA

ISBN: (纸本)9781424436767

Distributed source coding is one of the enabling technologies for efficient bandwidth utilization in wireless sensor networks and is consequently of great current interest. This paper studies its application to audio signals, using a transform weighted interleaved vector quantization (TWIN-VQ) framework and allowing a sensor node to passively receive and use information from neighboring sensors that is being transmitted to the common joint decoder. Specifically, it uses the linear predictor coefficients generated as side information by TWIN-VQ for one source to determine its frame-by-frame correlations with another source and then conditionally encode MDCT coefficients of the second source. Based on conditional entropy calculations, exploitation of this correlation results in potential improvements of 54-58% in coding efficiency. Using a multi-context adaptive arithmetic coder, an actual bitrate reduction of 38% is achieved.

关键词： audio coding Decoding Source coding Bandwidth Computational complexity Energy consumption Data mining Sensor systems Sensor arrays Wireless sensor networks

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 << < 29 30 31 32 33 34 35 36 37 38 > >>

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：