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检索条件"主题词=Linear Predictive Coding"

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Low Complexity Search Method for G.723.1 MP-MLQ Algorithm

Low Complexity Search Method for G.723.1 MP-MLQ Algorithm

引用

International Conference on Intelligent Systems Design and Applications (ISDA)

作者： R. S. Lin Y. C. Chen F. K. Chen Department of Computer Science and Information Engineering Southern Taiwan University Taiwan

The Multi-pulse Maximum Likelihood Quantization (MP-MLQ) algorithm with high complexity and high quality in its analysis-by-synthesis optimization has been used by many speech coding standards. To reduce the computational complexity of MP-MLQ in ITU-T G.723.1 speech codec, in this paper, we propose a fast search algorithm by using a designed function and the multi-track positions structure to predict the candidate pulses for each subframe. Simulation results show that the degradation of PESQ score for the pro-posed approach is about 0.069 relative to the original search procedure. However, the proposed approach reduced the computational complexity about 98.346% with perceptually negligible degradation.

关键词： Search methods Stochastic processes Computational complexity Speech coding Internet telephony Quantization linear predictive coding Algorithm design and analysis Degradation Personal digital assistants

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GRID architecture through a public cluster

GRID architecture through a public cluster

引用

International Conference on Computer and Communication Engineering, ICCCE

作者： Z. Akbar L.T. Handoko Group for Theoretical and Computational Physics Research center for Physics Indonesian Institute of Sciences Tangerang Indonesia

An architecture to enable some blocks consisting of several nodes in a public cluster connected to different grid collaborations is introduced. It is realized by inserting a web-service in addition to the standard Globus Toolkit. The new web-service performs two main tasks : authenticate the digital certificate contained in an incoming requests and forward it to the designated block. The appropriate block is mapped with the user name of the blockpsilas owner contained in the digital certificate. It is argued that this algorithm opens an opportunity for any blocks in a public cluster to join various global grids.

关键词： linear predictive coding Resource management Parallel machines Grid computing Computer architecture Joining processes Middleware Physics computing Collaboration Clustering algorithms

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Diminishing learning based SVM classifier with non-linear kernels

Diminishing learning based SVM classifier with non-linear ke...

引用

International Conference on Electronic Design (ICED)

作者： J. Manikandan B. Venkataramani Department of ECE National Institute of Technology Trichy Tamil Nadu India

Support Vector Machine (SVM) is one of the state-of-the-art tools for linear and nonlinear pattern classification. One of the design issues in SVM classifier is reducing the number of support vectors without compromising the classification accuracy. In this paper, a novel technique known as Diminishing Learning (DL) is proposed for an SVM based multi-class pattern recognition system. In this technique, a sequential classifier is proposed wherein the classes which require stringent boundaries are tested one by one and once the tests for these classes fail, the stringency of the classifier is increasingly relaxed. The effect of, the sequence in which the classes are trained and tested, on the recognition accuracy is also studied in this paper. The proposed technique is applied for SVM based isolated digit recognition system and is studied using speaker dependent TI46 database of isolated digits. Two feature extraction techniques, one using LPC and another using MFCC are applied to the speech from the above database and the features are mapped using SOFM. This in turn is used by the SVM classifier to evaluate the recognition accuracy with and without DL technique. Based on this study, it is found that the use of diminishing learning reduces the number of support vectors by 41.54% and 44.70% respectively for SVM classifier with LPC and MFCC feature inputs respectively. Recognition accuracies of 97% and 98% are achieved for SVM classifier with and without DL technique for LPC feature inputs respectively. Recognition accuracy of 100% is achieved for SVM with and without DL technique for MFCC feature inputs. The study confirms the effect of, the order in which the classes are trained and tested, on the recognition accuracy and for the TI46 database, about 2% to 4% increase in recognition accuracy is obtained by choosing the optimum order.

关键词： Support vector machines Support vector machine classification Kernel Spatial databases linear predictive coding Mel frequency cepstral coefficient Sequential analysis Testing Pattern classification Pattern recognition

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A Finite State Vector Quantizer for New Bit Rate Speech Compression

A Finite State Vector Quantizer for New Bit Rate Speech Comp...

引用

International Conference on Signal Processing, Communication and Networking (ICSCN)

作者： O. Wiriyanuruknakorn J. Srinonchat Electronics and Telecommunication Engineering Department Faculty of Engineering and Architecture Rajamangala University of Technology Nonthaburi Thailand Signal and System Research Laboratory Electronics and Telecommunication Engineering Department Rajamangala University of Technology Thanyaburi Pathum Thani Thailand

This paper presents an enhancement technique to compress speech signal based on LPC-10 model. Line spectrum pairs (LSP), as alternate representation of LPC presents some useful properties for quantisation and transmission, is used to model the speech signal in each frame. Artificial Neural Networks, based on a finite state vector quantizer technique, is investigates the characteristic of LSP coefficients to reduce the bit rate of the system. This technique shows an efficiency reduction of bit rate. The results of this experiments show that this technique can reduce the bit rate approximately 5.5 -37% and also provides the excellent formant frequency similar to the original speech.

关键词： Bit rate linear predictive coding Quantization Speech synthesis predictive models Speech enhancement Frequency Humans Space technology Speech coding

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A Precise Estimation of Vocal Tract Parameters for High Quality Voice Morphing

A Precise Estimation of Vocal Tract Parameters for High Qual...

引用

9th International Conference on Signal Processing(第九届国际信号处理学术会议)(ICSP'08)

作者： Ning Xu Zhen Yang Institute of Signal Processing and Transmission Nanjing University of Post and Telecommunication NanjingChina

One of the most recent models for voice conversion is the classical LPC analysis-synthesis model combined with GMM,which aims to separate information from excitation and vocal tract and to learn the transformation rules with statistical ***,it does not work well as it is supposed to be due to the inaccuracy of the extracted feature information as well as the overly-smoothed spectral converted by traditional *** this paper,we propose a novel method to solve the problem which is based on the technique of the separation of glottal waveforms and the prediction of the *** final result shows that not only are the transformed vocal tract parameters matching the target one better,but also is the high quality of the synthesized speech preserved.

关键词： linear predictive coding Speech synthesis Speech analysis Feature extraction Signal synthesis Signal processing Information analysis Signal analysis Statistical analysis Data mining

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An Approach to Information Hiding in Low Bit-Rate Speech Stream

An Approach to Information Hiding in Low Bit-Rate Speech Str...

引用

IEEE GLOBECOM 2008 - 2008 IEEE Global Telecommunications Conference

作者： Bo Xiao Yongfeng Huang Shanyu Tang Dept. of Electron. Eng. Tsinghua Univ. Tsinghua Nat. Lab. for Inf. Sci. & Technol. Beijing Dept. of Computing London Metropolitan University London UK

In this paper we introduce a novel codebook partition algorithm for quantization index modulation (QIM), which is applied to information hiding in instant low bit-rate speech stream. The QIM method divides the codebook into two parts, each representing '0' and '1' respectively. Instead of randomly partitioning the codebook, the relationship between codewords is considered. The proposed algorithm - complementary neighbor vertices (CNV) guarantees that every codeword is in the opposite part to its nearest neighbor, and the distortion is limited by a bound. The feasibility of CNV is proved with graph theory. Moreover, in our work the secret message is embedded in the field of vector quantization index of LPC coefficients, getting the benefit that the distortion due to QIM is lightened adaptively by the rest of the encoding procedure. Experiments on iLBC and G.723.1 verify the effectiveness of the proposed method. Both objective and subjective assessments show the proposed method only slightly decreases the speech quality to an indistinguishable degree. The hiding capacity is no less than 100 bps. To the best of our knowledge, this is the first work adopting graph theory to improve the codebook partition while using QIM in low bit-rate streaming media.

关键词： linear predictive coding Filters Partitioning algorithms Graph theory Nearest neighbor searches Vector quantization Speech coding Laboratories Information science Modulation coding

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Tracking In the Presence of a Time-Varying Uncertain Gain

Tracking In the Presence of a Time-Varying Uncertain Gain

引用

Annual Allerton Conference on Communication, Control, and Computing

作者： Julie R. Vale Daniel E. Miller Dept. of Electrical and Computer Engineering University of Waterloo Waterloo Ontario CANADA

ISBN: (纸本)9781424429257

In this paper we consider the use of linear periodic controllers to solve a tracking problem in the face of time-varying multiplicative gain uncertainty. The class of signals to be tracked is modelled by a (stable) filter on the reference input. We show, under the technical assumption that the relative degree of the filter is greater than the plant, that there exists a linear periodic controller which can provide near-optimal linear time-invariant performance for every admissible gain.

关键词： Uncertainty Filters linear predictive coding Underwater tracking Performance gain Feedback Stability Size control Size measurement Switches

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NetFailPAC: A Single Failure Protection Algorithm with QoS Provision for Optical WDM Networks

NetFailPAC: A Single Failure Protection Algorithm with QoS P...

引用

ICONS International Conference on Systems

作者： W. Grote A. Arenas A. Zapata Electrical and Electronics Engineering Universidad Tècnica Federico Santa María Chile

By sharing the protection bandwidth of high- priority connections (HPC) with the bandwidth of low- priority (LPC) ones, NetFailPAC (network failure protection algorithm with class of service) establishes working (HPC and LPC) and protection routes (for HPC only) in a WDM network. By using bandwidth as the main metric to select the working and backup paths, it outperforms previous proposals increasing high-priority connection setup capabilities from 0% to 50% without additional infrastructure, while maintaining similar network performance.

关键词： Protection Optical fiber networks WDM networks Bandwidth linear predictive coding Quality of service Spine Proposals Degradation Wavelength division multiplexing

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DISTANT-TALKING ROBUST SPEECH RECOGNITION USING LATE REFLECTION COMPONENTS OF ROOM IMPULSE RESPONSE

DISTANT-TALKING ROBUST SPEECH RECOGNITION USING LATE REFLECT...

引用

2008 IEEE International Conference on Acoustics, Speech and Signal Processing, Pages 4521-4972

作者： Randy Gomez Jani Even Hiroshi Saruwatari Kiyohiro Shikano Graduate School of Information Science Nara Institute of Science and Technology Japan

We propose a robust and fast dereverberation technique for real-time speech recognition application. First, we effectively identify the late reflection components of the room impulse response. We use this information together with the concept of Spectral Subtraction (SS) to remove the late reflection components of the reverberant signal. In the absence of the clean speech in actual scenario, approximation is carried out in estimating the late reflection where the estimation error is corrected through multi-band SS. The multi-band coefficients are optimized during offline training and used in the actual online dereverberation. The proposed method performs better and faster than the relevant approach using Multi-LPC and reverberant matched model. Moreover the proposed method is robust to speaker and microphone locations.

关键词： Robustness Speech recognition Reflection Reverberation linear predictive coding Information science Estimation error Microphones Degradation Equations

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Improving audio data quality and compression

Improving audio data quality and compression

引用

International Conference on Emerging Technologies, ICET

作者： Qaiser Naeem Muhammad Sharif Mudassar Raza Department of Computer Science COMSATS Institute of Information Technology Punjab India

High data quality at low bit rate is an essential goal that people want to achieve. It is necessary to transfer data at low bit rate so that the bandwidth of the medium can be utilized efficiently. In most of the speech coding techniques the goal of low bit rate transfer is achieved but the data quality is affected badly. The proposed technique is an attempt to improve the data quality at low bit rate as well as fast transmission of data. The proposed technique protects the data quality by applying linear predictive coding-10 and achieves the low bit rate by applying Quadrature Mirror Filter. A comprehensive analysis is on the basis of given parameters as size, compression time, Signal to Noise Ratio, power, energy, power in air, energy in air, mean, standard deviation and intensity.

关键词： Speech coding Bit rate Pulse modulation linear predictive coding Nonlinear filters Modulation coding Narrowband Digital modulation predictive models Bandwidth

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