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351-360 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：

A robust speech/music discriminator for switched audio coding

A robust speech/music discriminator for switched audio codin...

引用

European Signal Processing Conference (EUSIPCO)

作者： Guillaume Fuchs Fraunhofer lnstitut für Integrierte Schaltungen (IIS) Germany

Switching between speech coding and generic audio coding schemes was recently proven to be very efficient for coding a large range of audio materials at low bit-rates. However, it strongly relies on a robust classification of the input signal. The aim of the paper is to design a reliable speech and music discriminator (SMD) for such an application. Main attention was laid on getting a good tradeoff between accuracy, reactivity and stability of the decision while keeping the delay and complexity reasonably low. To this end, short-term and long-term features are dissociated before being conveyed to two different classifiers. The two classifier outputs are combined in a final decision using a hysteresis. Objective measures show that a more reliable switching decision is achievable. The SMD was successfully implemented in MPEG Unified Speech and audio coding (USAC). It allows the codec to show unprecedented audio quality.

关键词： Speech Speech coding Switches audio coding Delays Feature extraction

来源：评论

学校读者我要写书评

暂无评论

Improved lossless audio coding using the noise-shaped IntMDCT

Improved lossless audio coding using the noise-shaped IntMDC...

引用

IEEE Digital Signal Processing Workshop

作者： Y. Yokotani R. Geiger G. Schuller S. Oraintara K.R. Rao Electrical Engineering Department University of Texas Arlington Arlington TX USA Fraunhofer Institute for Digital Media Technology Ilmenau Germany

This paper discusses approximation noise shaping to improve the efficiency of the integer modified discrete cosine transform (IntMDCT)-based lossless audio codec. The scheme is applied to rounding operations associated with lifting steps to shape the noise spectrum towards the low frequency bands. In this paper, constraints on the noise shaping filter and a design procedure with the constraints are discussed. Several noise shaping filters are designed and experimental results showing the improvement are presented.

关键词： audio coding Noise shaping Low-frequency noise Frequency Noise cancellation Filters Roundoff errors Floors Noise level Discrete cosine transforms

来源：评论

学校读者我要写书评

暂无评论

Scalable audio coding Based on Integer Transform

Scalable Audio Coding Based on Integer Transform

引用

International Conference on Communications and Networking in China (CHINACOM)

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

The modified discrete cosine transform (MDCT) is widely used in current perceptual audio coding schemes. This paper presents an integer approximation of this lapped transform, called integer MDCT, which is derived from the MDCT using lifting scheme. This reversible integer transform inherits most of the attractive properties of the MDCT, exhibiting a good spectral representation of the audio signal, critical sampling and overlapping, which makes the integer MDCT well suited for both lossless audio coding as well as for combined perceptual and lossless audio coding. A scalable system based on MPEG AAC and integer MDCT is presented providing a lossless enhancement of perceptual audio coding scheme.

关键词： audio coding Discrete transforms Discrete cosine transforms Sampling methods Prototypes Matrix decomposition Predictive coding Transform coding Image reconstruction Frequency

来源：评论

学校读者我要写书评

暂无评论

Low bitrates audio coding using lattice vector quantization based on scalable and high order codebook extension scheme

Low bitrates audio coding using lattice vector quantization ...

引用

International Conference on Wireless Communications and Signal Processing (WCSP)

作者： Kai Xiang Ruimin Hu National Engineering Research Center for Multimedia Software Wuhan University School of Computing Hubei University of Economics Wuhan China National Engineering Research Center for Multimedia Software Wuhan University of China Wuhan China

In the paper, we describe a lattice vector quantization model based on scalable and high order codebook extension scheme. In the scheme, we expand the basic codebook so that the expanded codebook contains the point to be quantized, and then fractionize the expanded Voronoi cell, and use the subdivided lattice point to quantize the point, thereby reducing increasing quantization distortion caused by cell-amplification. By repeating such a 'cell extension - cell subdivision - vector quantization' process, we can gradually increase the quantization precision of outside points and achieve progressively refined quantization at each stage of codebook extension. We demonstrate the efficiency of the proposed scheme in the application of speech spectrum quantization. Experiments on encoding multiple low bitrates audio sequences show that the scheme provides superior performance with moderate increase in complexity when compared with conventional lattice vector quantization by Voronoi extension used in AMR-WB+ audio codec.

关键词： Lattices Vector quantization Bit rate Signal to noise ratio audio coding

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

Multiple Description audio coding for Wireless Low-Frequency Sound Zones

Multiple Description Audio Coding for Wireless Low-Frequency...

引用

Data Compression Conference (DCC)

作者： Jan Østergaard Christian Pedersen Mo Zhou Niels de Koeijer Martin Møller Aalborg University Fredrik Bajers Vej 7B Aalborg Denmark TBang & Olufsen Bang og Olufsen Alle Struer Denmark

We present a joint design of sound zone control filters and robust audio coding for wireless low frequency sound zones. The audio signal is filtered using sound zone control filters and encoded using a multiple-description coder. The control filters and the multiple-description coder are combined in a nested loop. The inner loop performs filtering for sound zone control and generates multiple descriptions using oversampling and closed-loop prediction. The outer loop performs noise shaping and guarantees a trade-off between robustness and quality of the descriptions. A closed-form expression for the optimal sound-zone control filters are provided, and a simulation study demonstrates that even at moderate packet loss rates, a significant gain is possible compared to not using multiple descriptions.

关键词： Wireless communication Closed-form solutions Finite impulse response filters audio coding Filtering Packet loss Robustness

来源：评论

学校读者我要写书评

暂无评论

A rapidly adaptive lossless compression algorithm for high fidelity audio coding

A rapidly adaptive lossless compression algorithm for high f...

引用

Data Compression Conference (DCC)

作者： T. Shamoon C. Heegard School of Electrical Engineering Cornell University Ithaca NY USA

A new adaptive algorithm for lossless compression of digital audio is presented. The algorithm is derived from ideas from both dictionary coding and source-modeling. An adaptive Lempel-Ziv (1977) style fixed dictionary coder is used to build a source model that fuels an arithmetic coder. As a result, variable length strings drawn from the source alphabet are mapped onto variable length strings that are on average shorter. The authors show that this algorithm outperforms arithmetic coding or Lempel-Ziv coding working alone on the same source (in their experiments the source is an ADPCM quantizer). Adaptation heuristics for the Lempel-Ziv coder, relevant data structures, and a discussion of audio source modeling (entropy estimation) experiments are described. While the algorithm presented herein is designed to be used as a post-compressor in a lossy audio transform coding system, it is well suited for any instance where non-stationary source outputs must be compressed.< >

关键词： Compression algorithms audio coding Entropy Bit rate audio compression Frequency Adaptive algorithm Dictionaries Arithmetic Transform coding

来源：评论

学校读者我要写书评

暂无评论

ENHANCEMENT OF PRINCIPAL TO AMBIENT ENERGY RATIO FOR PCA-BASED PARAMETRIC audio coding

ENHANCEMENT OF PRINCIPAL TO AMBIENT ENERGY RATIO FOR PCA-BAS...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Se-Woon Jeon Dongil Hyun Jeongil Seo Young-Cheol Park Dae-Hee Youn School of Electrical & Electronic Eng. Yonsei Univ. Seoul Korea Broadcasting Media Technology Department ETRI Daejeon Korea Computer & Telecommunications Eng. Division Yonsei Univ. Wonju Gangwon Korea

ISBN: (纸本)9781424442959;9781424442966

In this paper, we propose a method of restoring principal to ambient energy ratio (PAR) at the decoder in the principal component analysis (PCA)-based parametric audio coding. The conventional approach applies the post-scaling at the decoder using the energy information extracted from the input signal at the encoder. However, this approach has a problem that the relative energy of principal source in the reconstructed signal is smaller than the original signal and also affected by the panning angle of principal source. To restore the PAR at the decoder, the proposed method estimates the post-scaling factors using parametric information extracted from PCA-based formulation. The objective and subjective results verify the performance improvement of proposed method.

关键词： Parametric audio coding PCA stereo audio coding principal components analysis decoders Passive Cutaneous Anaphylaxis Energy information audio Energy sources ambient environment input signal Visible Light JTB gene performance boost original signal principal Principal

来源：评论

学校读者我要写书评

暂无评论

A scalable subband audio coding scheme for ATM environments

A scalable subband audio coding scheme for ATM environments

引用

IEEE Southeastcon

作者： Wen Jin M.S. Scordilis Department of Electrical and Computer Engineering University of Miami Coral Gables FL USA

ISBN: (纸本)0780367480

A scalable audio coder is proposed for maintaining uninterrupted service in dynamically changing ATM network environments. Performance requirements include seamless switching between bit rates, which would gracefully vary the perceived signal quality. The encoder is similar to the MPEG-1 Layer 3 (MP3) standard but differs significantly in that the MDCT coefficients are quantized in a layered manner. Four enhancement layers that progressively address residual quantization errors follow a perceptually optimized core layer with somewhat poor quantization resolution. Each layer generates its own independently decodable bit stream. Core layer bits appended with enhancement bits are packed into one frame. Signal quality gracefully varies from basic to transparent with the corresponding bit rates of 8, 16, 32, 64, and 128 kb/s at each layer. Subjective tests indicate that our codec performs equivalently to MP3 at 128 kb/s with a somewhat inferior performance at lower bit rates.

关键词： audio coding Bit rate Asynchronous transfer mode Digital audio players Quantization Signal resolution Decoding Testing Codecs Performance evaluation

来源：评论

学校读者我要写书评

暂无评论

Ultra-low delay lossy audio coding using DPCM and block companded quantization

Ultra-low delay lossy audio coding using DPCM and block comp...

引用

Australian Communications Theory Workshop (AusCTW)

作者： Gediminas Simkus Martin Holters Udo Zölzer Department of Signal Processing and Communications Helmut Schmidt University-University of the Federal Armed Forces Hamburg Hamburg Germany

Real-time audio transmission requires good quality for a restricted channel capacity and minimum latency. An ultra-low delay audio coding scheme based on differential pulse code modulation (DPCM) and block companded quantization is presented. The prediction filter of the base backward DPCM codec is attained as a FIR filter in lattice structure. The proposed block based audio coding introduces an algorithmic latency below one millisecond and an overhead of less than a half bit per sample. The advantage of the block companded quantization with DPCM is the capability to follow rapid changes in the signal. Therefore, it significantly improves the perceptual audio quality compared to a plain DPCM coding scheme with an adaptive quantizer.

关键词： Lattices Quantization (signal) Decoding Delays audio coding Speech

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 << < 32 33 34 35 36 37 38 39 40 41 > >>

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：