检索结果-内蒙古大学图书馆

A 6.7 kbps vector sum excited linear prediction on TMS320C54X digital signal processor

MICROPROCESSORS AND MICROSYSTEMS 2002年第1期26卷 27-38页

作者： Kumar, KND Prabhu, KMM Sommen, PCW Indian Inst Technol Dept Elect Engn Madras 600036 Tamil Nadu India Eindhoven Univ Technol Dept Elect Engn NL-5600 MB Eindhoven Netherlands

In this paper, a 6.7-kbps vector sum excited linear prediction (VSELP) coder with less computational complexity is presented. A very efficient VSELP codebook with nine basis vectors and a heuristic K-selection method (to reduce the search space and complexity) is constructed to obtain the stochastic codebook vector. The nine basis vectors are obtained by optimizing a set of randomly generated basis vectors. During the optimization process, we have trained the basis vectors to give the system apriori knowledge of the characteristics of the input. The coder is implemented on a TMS320C541 digital signal processor. The performance is evaluated by testing the 6.7-kbps VSELP coder with different test speech data taken from different speakers. The quality of the coder is estimated by comparing the performance of the 6.7-kbps VSELP coder with an 8-kbps VSELP speech coder based on the IS-54 standards. (C) 2002 Elsevier Science B.V. All rights reserved.

关键词： vector sum excited linear prediction code excited linear prediction linear predictive coding digital signal processor

来源：评论

学校读者我要写书评

暂无评论

SHORT-TERM TEMPORAL DECOMPOSITION AND ITS PROPERTIES FOR SPEECH COMPRESSION

引用

IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第6期39卷 1282-1290页

作者： CHENG, YM OSHAUGHNESSY, D MCGILL UNIV DEPT ELECT ENGNMONTREAL H3A 2T5QUEBECCANADA

This paper deals with a technique of temporal speech compression, called temporal decomposition. A mathematical presentation of temporal decomposition is discussed from the point of view of speech production. We theoretically study temporal decomposition in terms of an iterative solution and its convergence. For vocoder applications, we propose a practical algorithm, short-term temporal decomposition. The statistical performance of this algorithm was tested on a multispeaker database. We comparatively study the efficiencies of vector quantization and vector predictability on data derived from the algorithm and on data from LPC analysis. The results show that data from the algorithm have more efficient information transmission via vector quantization than data from LPC, and that vector prediction coding is feasible for the data from temporal decomposition.

关键词： Speech coding Vocoders Iterative algorithms Vector quantization linear predictive coding Speech analysis Interpolation Matrix decomposition Distortion measurement Testing

来源：评论

学校读者我要写书评

暂无评论

Joint quantisation strategies for low bit-rate sinusoidal coding

引用

IET SIGNAL PROCESSING 2010年第5期4卷 548-559页

作者： Unver, E. Villette, S. Kondoz, A. Univ Surrey Ctr Commun Syst Res Guildford GU2 7XH Surrey England

Although there are speech coding standards producing high-quality speech above 4 kbps, below that transparent quality has not been achieved yet. There is still room for improvement at lower bit rates, especially at 2.4 kbps and below, which is an area of interest for military and security applications. Strategies for achieving high-quality speech using sinusoidal coding at very low bit rates are discussed. Previous work in the literature on combining several frames in a metaframe and performing variable bit allocation within the metaframe is extended. Experiments have been carried out to find an optimum metaframe size compromise between delay and quantisation gains. Metaframe classification and quantisation according to the metaframe class are used for better efficiency. A method for voicing determination from the linear prediction coefficient (LPC) shape is also presented. The proposed techniques have been applied to the SB-LPC vocoder to produce speech at 1.2 and 0.8 kbps, and compared to the original SB-LPC vocoder at 2.4/1.2 kbps as well as an established standard (Mixed Excitation linear predictive - MELP - vocoder) at 2.4/1.2/0.6 kbps in a listening test. It has been found that the proposed techniques have been effective in reducing the bit rate while not compromising the speech quality.

关键词： low bit-rate sinusoidal coding metaframe classification speech coding Speech processing techniques speech quality linear prediction coefficient linear predictive coding Speech and audio coding mixed excitation linear predictive vocoder SB-LPC vocoder metaframe quantisation strategies Codecs, coders and decoders vocoders

来源：评论

学校读者我要写书评

暂无评论

GLOTTAL INVERSE FILTERING BY JOINT ESTIMATION OF AN AR SYSTEM WITH A linear INPUT MODEL

引用

IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1986年第1期34卷 28-42页

作者： MILENKOVIC, P WAISMAN CTR MENTAL RETARDAT & HUMAN DEV SPEECH MOTOR CONTROL LABSMADISONWI

Glottal inverse filtering is a technique by which the flow past the time varying glottal constriction is estimated by a filtering operation on the acoustic signal in human speech. The filtering operation removes the effects of the vocal tract resonances to reveal the underlying voice source signal. A linear model of a glottal pulse waveform is described along with a procedure for jointly determining an AR model of the vocal tract response together with the parameters of the glottal pulse model. This technique is applied to inverse filtering in human speech and the results are compared to covariance method LPC analysis.

关键词： Nonlinear filters Filtering Speech analysis Pulse shaping methods Speech synthesis Shape Resonance linear predictive coding Algorithm design and analysis Humans

来源：评论

学校读者我要写书评

暂无评论

A FEEDBACK-CONTROL SYSTEM FOR REAL-TIME FORMANT ESTIMATION .2. ANALYSIS OF A HYSTERESIS EFFECT AND F2 ESTIMATION

引用

IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1993年第9期40卷 892-898页

作者： ZIERHOFER, CM HOCHMAIR, ES Institut für Experimentalphysik Universität Innsbruck Innsbruck Austria

This paper presents the second part of the analysis of a feedback control system for real-time formant estimation. The system behavior is analyzed for an input signal composed of two sinusoids. If the frequency difference between the two input spectral lines is sufficiently great and the amplitude ratio is within certain limits, a hysteresis effect occurs. Then the system shows a tendency to select one of the two input spectral lines. The existence of the second line has only little influence on the accuracy of the detection of the selected line. From the analysis, conclusions of the system behavior regarding formant estimation can be drawn. A design example for second formant detection is simulated and compared with the results obtained by simulation of a zero-crossing system for F2 estimation and LPC analysis.

关键词： Feedback control Real time systems Hysteresis Analytical models predictive models linear predictive coding Signal analysis Frequency

来源：评论

学校读者我要写书评

暂无评论

Study of forecasting renewable energies in smart grids using linear predictive filters and neural networks

引用

IET RENEWABLE POWER GENERATION 2011年第6期5卷 470-480页

作者： Moghaddam, A. Anvari Seifi, A. R. Shiraz Univ Dept Power & Control Sch Elect & Comp Engn Engn Fac 1 Shiraz Iran

Accurate forecasting of renewable energies such as wind and solar has become one of the most important issues in developing smart grids. Therefore introducing suitable means of weather forecasting with acceptable precision becomes a necessary task in today's changing power world. In this work, an intelligent way for hourly estimation of both wind speed and solar radiation in a typical smart grid has been proposed and its superior performance is compared to those of conventional methods and neural networks (NNs). The methodology is based on linear predictive coding and digital image processing principles using two dimensional (2-D) finite impulse response filters. Meteorological data have been collected during the period 1 January 2009 to 31 December 2009 from Casella automatic weather station (AWS) at Plymouth, UK. Numerical results indicate that a considerable improvement in forecasting process is achieved with 2-D predictive filtering compared to the conventional approaches.

关键词： Other power apparatus and electric machines linear predictive filters meteorological data power engineering computing Energy resources two dimensional finite impulse response filters Neural computing techniques weather forecasting wind speed Power engineering computing smart grids Energy resources and fuels linear codes neural networks neural nets solar radiation smart power grids renewable energies forecasting digital image processing principles power filters Casella automatic weather station 2D FIR filters linear predictive coding FIR filters renewable energy sources

来源：评论

学校读者我要写书评

暂无评论

Design and implementation of a parametric speech coder

引用

IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1998年第1期44卷 163-169页

作者： Kwong, S Nui, PT City Univ Hong Kong Dept Comp Sci Hong Kong Hong Kong

Currently, it has been found out that speech coders based on linear predictive coding are good and accurate in modeling speech utterances. It does not only modeled speech utterance in an accurate manner but it also has the following properties: i) it provides a good estimate of the vocal tract spectral envelope;ii) it is analytically tractable;iii) it can be implemented in either software or hardware;iv) it uses lesser data storage than many other approaches of speech coders. In this paper, a hybrid speech coder based on Code Excited linear predictive coding (CELPC) and Voice Excited linear predictive coding (VELPC) is presented. CELPC is currently one of the main techniques for producing high quality speech at around 4.8 kbps. However, the computation requirement for the original CELPC was too demanding and not suitable for practical use. Therefore, a hybrid approach to produce speech signals with high quality and with a bit rate of 3.478 kbps has been proposed. This method splits the speech signal into two portions, the base-band signal and the high-band signal in the frequency domain. The base-band and the high-band signal are then coded using the CELPC and VELPC techniques, respectively.

关键词： Speech coding linear predictive coding Signal processing Speech analysis Nonlinear filters predictive models Bit rate Quantization Encoding Computer science

来源：评论

学校读者我要写书评

暂无评论

MAXIMUM-LIKELIHOOD-ESTIMATION OF THE PARAMETERS OF DISCRETE FRACTIONALLY DIFFERENCED GAUSSIAN-NOISE PROCESS

引用

IEEE TRANSACTIONS ON SIGNAL PROCESSING 1993年第10期41卷 2977-2990页

作者： DERICHE, M TEWFIK, AH Department of Electrical Engineering University of Minnesota Minneapolis MN USA

A maximum likelihood estimation procedure is constructed for estimating the parameters of discrete fractionally differenced Gaussian noise from an observation set of finite size N. The procedure does not involve the computation of any matrix inverse or determinant. It requires N2/2 + O(N) operations. The expected value of the loglikelihood function for estimating the parameter d of fractionally differenced Gaussian noise (which corresponds to a parameter of the equivalent continuous-time fractional Brownian motion related to its fractal dimension) is shown to have a unique maximum with the range of allowable values of d. The maximum occurs at the true value of d. A Cramer-Rao bound on the variance of any unbiased estimate of d obtained from a finite size observation set is derived. It is shown experimentally that the maximum likelihood estimate of d is unbiased and efficient when finite size data sets are used in the estimation procedure. The proposed procedure is also extended to deal with noisy observations of discrete fractionally differenced Gaussian noise.

关键词： Maximum likelihood estimation Gaussian noise Parameter estimation Brownian motion Fractals Signal processing Speech processing linear predictive coding Filters Motion estimation

来源：评论

学校读者我要写书评

暂无评论

Time-Varying Feature Selection and Classification of Unvoiced Stop Consonants

引用

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第3期2卷 395-405页

作者： Nathan, Krishna S. Silverman, Harvey F. Brown Univ Div Engn LEMS Providence RI 02912 USA

A feature set that captures the dynamics of formant transitions prior to closure in a VCV environment is used to characterize and classify the unvoiced stop consonants. The feature set is derived from a time-varying, data-selective model for the speech signal. Its performance is compared with that of comparable formant data from a standard delta-LPC-based madel. The different feature sets are evaluated on a database composed of eight talkers. A 40% reduction in classification error rate is obtained by means of the time-varying model. The performance of three different classifiers is discussed. A novel adaptive algorithm, termed learning vector classifier (LVC) is compared with standard K-means and LVQ2 classifiers. LVC is a supervised learning classifier that improves performance by increasing the resolution of the decision boundaries. Error rates obtained for the three-way (p, t, and k) classification task using LVC and the time-varying analysis are comparable to that of techniques that make use of additional discriminating information contained in the burst. Further improvements are expected when an expanded time-varying feature set is utilized, coupled with information from the burst.

关键词： Speech Error analysis linear predictive coding Spatial databases Adaptive algorithm Supervised learning Information analysis Lips Teeth

来源：评论

学校读者我要写书评

暂无评论

R/D optimal linear prediction

引用

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2000年第6期8卷 646-655页

作者： Prandoni, P Vetterli, M Ecole Polytech Fed Lausanne Lab Commun Audio Visuelle CH-1015 Lausanne Switzerland Univ Calif Berkeley Dept Elect Engn & Comp Sci Berkeley CA 94720 USA

A common technique to extend linear prediction to nonstationary signals is time segmentation: the signal is split into small portions and the modelization is carried out locally, The accuracy of the analysis is, however dependent on the window size and on the signal characteristics, so that the problem of finding a good segmentation is crucial to the entire modeling scheme, In this paper, we will present an algorithm which determines the optimal segmentation with respect to a cost function relating prediction error to modeling cost. The proposed approach casts the problem in a rate/distortion (R/D) framework, whereby the segmentation is implicitly computed while minimizing the modelization distortion for a given modelization cost. The algorithm is implemented by means of dynamic programming and takes the form of a trellis-based Lagrangian minimization, The optimal linear predictor, when applied to speech coding, dramatically reduces the number of bits per second devoted to the modeling parameters in comparison to fixed-window schemes.

关键词： dynamic programming linear predictive coding nonstationary signal segmentation rate-distortion

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：