检索结果-内蒙古大学图书馆

54th IEEE International Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Mabrouk, Ahmed Hassim, Norhidayah IIUM Dept Comp Sci Kl Malaysia

ISBN: (纸本)9781612848570

Line Spectral Frequencies (LSF) have been widely used to represent Linear Prediction Coefficients (LPC) due to their excellent quantization characteristics. Among others, MPEG4-compliant audio decoders include a module for converting the LSF to LPC coefficients. This paper presents a new simplified computational technique for carrying out the LSF-to-LPC conversion process. The proposed technique cuts down on several arithmetic operations by establishing direct links between the LPC coefficients and the two polynomials representing the two extreme glottis conditions. The technique is a good candidate for hardware implementation on mobile devices designed for low bit-rate wireless communication. Results of two testing scenarios are reported to show the computational distortion of the hardware conversion module relative to that resulting from quantizing the LSF vectors.

关键词： Distortion measurement Indexes LSF-based speech decoder LSF-to-LPC conversion MPEG4-compliant audio decoders arithmetic operations audio coding bit-rate wireless communication computational distortion glottis conditions hardware conversion module line spectral frequencies linear prediction coefficients mobile devices polynomials polynomials quantisation (signal) quantization characteristics simplified computational technique speech coding

来源：评论

学校读者我要写书评

暂无评论

A PERCEPTUALLY REWEIGHTED MIXED-NORM METHOD FOR SPARSE APPROXIMATION OF audio SIGNALS

A PERCEPTUALLY REWEIGHTED MIXED-NORM METHOD FOR SPARSE APPRO...

引用

45th Asilomar Conference on Signals, Systems and Computers (ASILOMAR)

作者： Christensen, Mads Graesboll Sturm, Bob L. Aalborg Univ Dept Architecture Design & Media Technol Aalborg Denmark

ISBN: (纸本)9781467303231

In this paper, we consider the problem of finding sparse representations of audio signals for coding purposes. In doing so, it is of utmost importance that when only a subset of the present components of an audio signal are extracted, it is the perceptually most important ones. To this end, we propose a new iterative algorithm based on two principles: 1) a reweighted 1-norm based measure of sparsity;and 2) a reweighted 2-norm based measure of perceptual distortion. Using these measures, the considered problem is posed as a constrained convex optimization problem that can be solved optimally using standard software. A prominent feature of the new method is that it solves a problem that is closely related to the objective of coding, namely rate-distortion optimization. In computer simulations, we demonstrate the properties of the algorithm and its application to real audio signals.

关键词： audio coding sparse approximations perceptual distortion measures audio modeling

来源：评论

学校读者我要写书评

暂无评论

Development of a temporal fundamental frequency coding strategy for cochlear implants

引用

JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2011年第6期129卷 4023-4036页

作者： Vandali, Andrew E. van Hoesel, Richard J. M. Hearing CRC Melbourne Vic 3053 Australia

A sound-coding strategy for users of cochlear implants, named enhanced-envelope-encoded tone (eTone), was developed to improve coding of fundamental frequency (F0) in the temporal envelopes of the electrical stimulus signals. It is based on the advanced combinational encoder (ACE) strategy and includes additional processing that explicitly applies F0 modulation to channel envelope signals that contain harmonics of prominent complex tones. Channels that contain only inharmonic signals retain envelopes normally produced by ACE. The strategy incorporates an F0 estimator to determine the frequency of modulation and a harmonic probability estimator to control the amount of modulation enhancement applied to each channel. The F0 estimator was designed to provide an accurate estimate of F0 with minimal processing lag and robustness to the effects of competing noise. Error rates for the F0 estimator and accuracy of the harmonic probability estimator were compared with previous approaches and outcomes demonstrated that the strategy operates effectively across a range of signals and conditions that are relevant to cochlear implant users. (C) 2011 Acoustical Society of America. [DOI: 10.1121/1.3573988]

关键词： acoustic signal processing audio coding cochlear implants

来源：评论

学校读者我要写书评

暂无评论

G.711.1 ANNEX D AND G.722 ANNEX B - NEW ITU-T SUPERWIDEBAND CODECS

G.711.1 ANNEX D AND G.722 ANNEX B - NEW ITU-T SUPERWIDEBAND ...

引用

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

作者： Miao, Lei Liu, Zexin Hu, Chen Eksler, Vaclav Ragot, Stephane Lamblin, Claude Kovesi, Balazs Sung, Jongmo Fukui, Masahiro Sasaki, Shigeaki Hiwasaki, Yusuke Huawei Technologies China VoiceAge Corp. Canada France Telecom Orange France Electronics and Telecommunications Research Institutes South Korea NTT Cyber Space Laboratories Japan

ISBN: (纸本)9781457705397

This paper presents high quality monaural superwideband extensions to G.711.1 and G.722, recently standardized as Recommendations ITU-T G.711.1 Annex D and G.722 Annex B. The superwideband (50-14000 Hz) functionality is achieved using embedded scalable structure that adds extension layers on top of the wideband core codecs. The bit rates are extended to 96/112/128 and 64/80/96 kbit/s for G.711.1 and G.722, respectively. The main technologies include lower and higher band (0-4 kHz and 4-8 kHz) enhancements, 8-14 kHz bandwidth extension and transform coding based on algebraic vector quantization. The codecs' performance is illustrated with listening test results extracted from formal ITU-T Characterization tests.

关键词： speech coding audio coding G.711.1 Annex D G.722 Annex B superwideband

来源：评论

学校读者我要写书评

暂无评论

GOSSET LATTICE SPHERICAL VECTOR QUANTIZATION WITH LOW COMPLEXITY

GOSSET LATTICE SPHERICAL VECTOR QUANTIZATION WITH LOW COMPLE...

引用

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

作者： Krueger, Hauke Geiser, Bernd Vary, Peter Li, Hai Ting Zhang, Deming Rhein Westfal TH Aachen Inst Commun Syst & Data Proc I D Aachen Germany

ISBN: (纸本)9781457705397

This paper introduces a novel and highly efficient realization of a spherical vector quantizer (SVQ), the "Gosset Low Complexity Vector Quantizer" (GLCVQ). The GLCVQ codebook is composed of vectors that are located on spherical shells of the Gosset lattice E-8. A high encoding efficiency is achieved by representing the spherical vector codebook as aggregated permutation codes. Compared to previous algorithms, the computational complexity and memory consumption is further reduced by exploiting the properties of so called classleader root vectors and by a novel approach for the codevector-to-index-mapping. The GLCVQ concept can be generalized to vector dimensions that are multiples of eight. In particular, GLCVQ for 16-dimensional vectors is used in Amd. 6 to ITU-T Rec. G.729.1.

关键词： Spherical vector quantization audio coding

来源：评论

学校读者我要写书评

暂无评论

Threshold Setting Model for the Set Partitioning in Hierarchical Trees (SPIHT) for audio

Threshold Setting Model for the Set Partitioning in Hierarch...

引用

International Conference on Graphic and Image Processing (ICGIP)

作者： Sondur, Shrinidhi Vilas Natl Inst Technol Karnataka Dept Elect & Commun Engn Surathkal India

ISBN: (纸本)9780819489326

This paper describes applying the Set Partitioning in Hierarchical Trees (SPIHT) algorithm using a devised model to compress audio signals. It allows choosing the amount of compression based upon bit rate requirements. A threshold setting model, based on energy and frequency patterns of the signal is used to assist the SPIHT encoder set efficient threshold values, based upon the nature of the audio. It is thus adaptable to the nature of the audio as well as the bit rate requirement. The implementation can be used for storage as well as progressive transmission.

关键词： Wavelet Threshold logic audio coding Sorting

来源：评论

学校读者我要写书评

暂无评论

EFFICIENT CONTEXT ADAPTIVE ENTROPY coding FOR REAL-TIME APPLICATIONS

EFFICIENT CONTEXT ADAPTIVE ENTROPY CODING FOR REAL-TIME APPL...

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Guillaume Fuchs Vignesh Subbaraman Markus Multrus Fraunhofer Institut Integrierte Schaltungen Am Wolfsmantel 33 91058 Erlangen Germany

ISBN: (纸本)9781457705380

Context based entropy coding has the potential to provide higher gain over memoryless entropy coding. However serious difficulties arise regarding the practical implementation in real-time applications due to its very high memory requirements. This paper presents an efficient method for designing context adaptive entropy coding while fulfilling low memory requirements. From a study of coding gain scalability as a function of context size, new context design and validation procedures are derived. Further, supervised clustering and mapping optimization are introduced to model efficiently the context. The resulting context modelling associated with an arithmetic coder was successfully implemented in a transform-based audio coder for real-time processing. It shows significant improvement over the entropy coding used in MPEG-4 AAC.

关键词： Entropy coding audio coding context modelling

来源：评论

学校读者我要写书评

暂无评论

Sound source localization based on B-format signals

Sound source localization based on B-format signals

引用

International Conference on Telecommunications and Signal Processing (TSP)

作者： Hasan Khaddour Department of Telecommunication FEEC Brno University of Technology Brno Czech Republic

This paper focuses on utilization of B-format signals in sound source localization using two different methods. Whereas the first method is based on the principle of B-format signals, the second one is based on energetic analysis of B-format signals. Average square difference function method is simulated and tested in order to be compared with the previous methods.

关键词： Microphones Delay effects Estimation Rendering (computer graphics) audio coding Azimuth Laboratories

来源：评论

学校读者我要写书评

暂无评论

Oversampled filter bank evaluation for joint subband audio processing and coding

Oversampled filter bank evaluation for joint subband audio p...

引用

Canadian Conference on Electrical and Computer Engineering (CCECE)

作者： David Hermann Edward Chau Robert D. Dony Shawki M. Areibi AMI Semiconductors Inc. Waterloo ONT Canada School of Engineering University of Guelph Guelph ONT Canada

This paper presents an experimental evaluation of oversampled, modulated filter banks for joint subband audio processing and coding applications. Joint subband processing and coding may be useful in some wireless audio devices such as advanced wireless digital hearing aids. We examine the use of oversampled GDFT and cosine modulated filter banks and propose using single sideband (SSB) real-valued filter banks as a compromise which is ideal for this application. The SSB filter bank provides real-valued signals for audio coding which are free from any aliasing cancellation constraints and hence are also suitable for audio processing such as subband gain adjustment. We support this conclusion with an experimental analysis of various filter bank designs for subband gain adjustment and subband audio coding.

关键词： Filter bank audio coding Delay Finite impulse response filter Hearing aids Modulation coding Amplitude modulation Prototypes Auditory system Ambient intelligence

来源：评论

学校读者我要写书评

暂无评论

A transient signal detection technique based on flatness measure

A transient signal detection technique based on flatness mea...

引用

International Conference on Computer Science & Education (ICCSE)

作者： Xuemin Zhang Changqing Cai Jianhong Zhang Faculty of Electrical&Information Engineering Changchun Institute of Technology Changchun China

In perceptual audio coding, it is necessary to deal with the transient signals in a frame. A reasonable transient detection is a premise of the treatment. According to the characteristics of transient signals in time domain and frequency domain, a time-frequency transient detection method which uses flatness measure is proposed in this paper. Comparing with existing transient detection methods, transient signal detection technique based on flatness measure not only has advantages to reduce missed and misused detection, but also hardly produce redundant detection in low-energy transient segment. Simultaneously, the complexity of the algorithm adaptively changes with the apparent extent of the transient signal. Simulation results show that this method is high in detection accurancy and simple in algorithm realization.

关键词： Transient analysis Frequency domain analysis Time domain analysis audio coding Frequency modulation

来源：评论

学校读者我要写书评

暂无评论

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：

请选择收藏分类：

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

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：