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Student Conference on Research and Development (SCOReD)

作者： A.M.M.A. Najih A.R. Ramli A. Ibrahim A.R. Syed Department of Computer and Communication System Engineering University Putra Malaysia Serdang Malaysia Department of Computer and Communication System Engineering University Putra Malaysia Serdang

Speech compression is one area of digital signal processing that focuses on reducing the bit rate of the speech signal for transmission or storage without significant loss of quality. In recent years a new technique called wavelet transform has been proposed for signal analysis. It has been successfully used in image compression application. So far, less attention has been paid to the research in the speech compression using wavelet. This paper attempts to evaluate the wavelet compression technique on speech signals. Different wavelet filters were used to select the best filter suitable for speech signal in providing low bit rate and low computation complexity. Our implementation was evaluated based on PSNR, SNR, NRMSE and compression ratio tested on 8 kHz 8-bit speech signals. This algorithm was also compared to the following speech compression schemes: linear predictive coding (LPC) which reduces the transmitted data by factor of more than twelve, and global system mobile (GSM) which reduces the transmitted data by factor of five As a result from this study, wavelet speech compression gives higher SNR and better speech quality than the other techniques.

关键词： Image coding Speech coding Speech processing Bit rate Speech analysis Filters linear predictive coding Digital signal processing Signal processing Propagation losses

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A real time spectral subtraction based speech enhancement scheme

A real time spectral subtraction based speech enhancement sc...

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Canadian Conference on Electrical and Computer Engineering (CCECE)

作者： D. Flogeras R. Doraiswami M.E. Kaye Department of Electrical and Computer En ineering University of New Brunswick Fredericton NB Canada

Various speech enhancement schemes are analyzed in terms of the conflicting real time requirements of computational delay, robustness and accuracy. A spectral subtraction scheme is found to be implementable in real time using the available digital processing board. The tasks such as the computation of the spectral contents of the speech plus noise and the noise, smoothing the estimates, and speech enhancement filtering are implemented in the frequency domain using FFT in view of the computational speed and robustness. A variant of a spectral subtraction scheme is implemented in real time on a DSP board and its performance is evaluated.

关键词： Speech enhancement predictive models Amplitude estimation Noise robustness Filtering Real time systems Frequency estimation linear predictive coding Niobium Speech analysis

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Vehicle license plate recognition in marketing application

Vehicle license plate recognition in marketing application

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IEEE Symposium on Intelligent Vehicle

作者： A. Taleb-Ahmed D. Hamad G. Tilmant LASL Université du Littoral Cote d'Opale Calais cedex LASL Université du Littoral Cote d'Opale Calais cedex France Heltys Roubaix Roubaix France

The paper presents an artificial vision system able to analyze the image of a car given by a camera, to locate the registration plate and to recognize its registration number. It describes practical problems encountered in implementing this application and the proposed solutions. The system has been designed using a modular approach. Sub-modules can be upgraded and or substituted independently, thus making the system potentially suitable in a large variety of vision applications. Performances of the system, have been evaluated in real situation.

关键词： Image recognition Cameras Licenses Image analysis Character recognition linear predictive coding Image processing Machine vision Image segmentation Vehicles

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Estimation of time-varying AR models of speech through Gauss-Markov modeling

Estimation of time-varying AR models of speech through Gauss...

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

作者： K.M. Malladi R.V. Rajakumar Hellosoft Inc. San Jose CA Dept. of E&ECE Indian Institute of Technology Kharagpur India

In this work, a new method for estimating the time-varying AR model of speech is presented. Here, the time-varying parameters are modeled as stationary processes. Both the time-varying parameters and their corresponding stationary process are modeled through a common Gauss-Markov model whose state-vector can be estimated through the extended Kalman Filter (EKF) algorithm. The proposed algorithm is different from the earlier methods which use the EKF algorithm. Simulation studies are carried out for both voiced and unvoiced speech. It is shown that the proposed method has less mean-square prediction error than that obtained through the LPC method.

关键词： Gaussian processes linear predictive coding Speech analysis Speech synthesis Signal synthesis Speech coding Parameter estimation State estimation Speech recognition Text recognition

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Near optimal LQR performance for a compact set of plants

Near optimal LQR performance for a compact set of plants

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IEEE Conference on Decision and Control

作者： D.E. Miller Department of Electrical and Computer Engineering Waterloo ONT Canada

Here we consider the problem of providing near optimal performance for a large set of possible models. We adopt the LQR framework in the single-input single-output (SISO) setting, and prove that given a compact set of controllable and observable plant models of a fixed order, we can construct a single linear periodic controller (LPC) which provides near optimal LQR performance.

关键词： Uncertainty Optimal control Robust control linear predictive coding Adaptive control Control systems Programmable control Costs Transfer functions State feedback

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Evaluation of a warped linear predictive coding scheme

Evaluation of a warped linear predictive coding scheme

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

作者： A. Harma Lab. of Acoustics & Audio Signal Processing Helsinki Univ. of Technol. Espoo Finland

Basically all conventional digital signal processing techniques can be warped by introducing a simple modification to the system. In this paper, the focus is in warped linear predictive coding techniques with application to speech and audio coding. The performance of warped LPC is compared with a conventional LPC in listening tests and in terms of technical measures. This is done at various sampling rates as a function of the order of the LPC model.

关键词： linear predictive coding Frequency Nonlinear filters Finite impulse response filter Transfer functions Convolution Equations Laboratories Acoustic signal processing Digital signal processing

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Correction to "A Compact, Flexible LPC Vocoder Based on a Commercial Signal Processing Microcomputer"

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IEEE Journal of Solid-State Circuits 2003年第3期18卷 376-376页

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

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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

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Unequal error protection and source-channel decoding of CELP speech

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ELECTRONICS LETTERS 2002年第7期38卷 347-349页

作者： Nazer, A Alajaji, F Queens Univ Dept Math & Stat Kingston ON K7L 3N6 Canada

The reliable communication of FS CELP 10 16 encoded speech over very noisy channels is investigated. Using second-order Markov chains it is shown that over one-quarter of the CELP bits in every frame of speech are redundant. An unequal error protection coding scheme, which exploits this residual redundancy, is proposed for sending the CELP parameters over Gaussian and Rayleigh fading channels. Simulations indicate substantial coding gains over conventional systems.

关键词： speech coding Gaussian channels combined source-channel coding reliable communication coding gains error correction codes voice communication decoding CELP speech second-order Markov chains linear predictive coding source-channel decoding Rayleigh channels telecommunication network reliability Markov processes unequal error protection coding FS CELP 1016 encoded speech redundancy residual redundancy very noisy channels Speech and audio coding Rayleigh fading channels

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Efficient implementation of perceptual postfilter for CELP coding of speech

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ELECTRONICS LETTERS 2002年第24期38卷 1604-1605页

作者： Chan, CF City Univ Hong Kong Dept Comp Engn & Informat Technol Hong Kong Hong Kong Peoples R China

An efficient method to implement the perceptual posterfilter for the suppression of coding noise in CELP-coded speech is proposed. The method is based on approximating the response of an all-pole filter to the response of the pole-zero form postfilter via cepstrum processing. This all-pole postfilter can then be implemented more efficiently than the pole-zero postfilter with less computation and filter memory.

关键词： poles and zeros 4.8 kbit/s noise all-pole filter response approximation code-excited linear predictive speech coders microprocessor speech coding digital filters perceptual postfilter coding noise suppression all-pole postfilter Digital filters pole-zero form postfilter decoding linear predictive coding CELP speech coding cepstrum processing filter memory CELP speech decoder Speech processing techniques Filtering methods in signal processing 8 bit cepstral analysis Speech and audio coding

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