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NEW METHOD FOR EXTRACTING SPEECH FORMANTS USING LPC PHASE SPECTRUM

ELECTRONICS LETTERS 1993年第24期29卷 2081-2082页

作者： CAI, JH JIANG, GJ ZHANG, LH Dept. of Electron. Eng. Hangzhou Univ. China

A new method for formant extraction using the LPC phase spectrum is proposed, which is especially effective in finding merged peaks. The bandwidth of a formant is easily calculated from the magnitude of the third deri... 详细信息

关键词： linear predictive coding SPEECH ANALYSIS AND PROCESSING

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Transmitter precoding with reduced-complexity soft detection for MIMO systems

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IEEE TRANSACTIONS ON WIRELESS COMMUNICATIONS 2007年第3期6卷 817-821页

作者： Li, Yong Moon, Jaekyun Univ Minnesota Dept Elect & Comp Engn Minneapolis MN 55455 USA

We present a precoded reduced-complexity soft detection (PRCSD) algorithm for multiple-input multiple-output (MIMO) systems. The linear operations at both transmit and receive sides based on complex Householder transform convert the MIMO channel to be multiple-diagonal, spatially partial-response-like, so that error propagation is alleviated when applying reduced-complexity soft detection at the receiver. The transform results in unitary precoding and feedforward matrices so that neither transmit power boost nor noise enhancement is present. Performance analysis based on pairwise error probability (PEP) shows that PRCSD achieves larger diversity advantage than existing precoding and multiple-beamforming (MB) schemes, which basically attempt to transmit signals through diagonal independent sub-channels and thus may suffer a diversity loss. PRCSD can achieve full diversity as maximum likelihood (ML) detection in some scenarios while reducing complexity significantly.

关键词： linear predictive coding

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MARKOV-HUFFMAN coding OF LPC PARAMETERS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1985年第2期33卷 451-453页

作者： PAPAMICHALIS, PE Central Research Laboratories Texas Instruments Inc. Dallas TX USA

The methods of Markov modeling and Huffman coding are combined to reduce the data rate of LPC-coded speech without any effect on the speech quality. A Markov model is applied to the quantization levels of the LPC parameters, and the transition probabilities are then used to generate Huffman coding tables. This procedure does not have any impact on speech quality since it affects only the representation of the quantized parameters. It is demonstrated that Markov-Huffman coding can lead to average savings of more than 20 percent in bit rate. A suboptimal scheme is also investigated, which can facilitate the implementation of the method on currently available signal processing chips.

关键词： linear predictive coding Huffman coding Speech coding Bit rate Probability Quantization Signal processing Vocoders Frequency Databases

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Improved pulsed residual excited linear prediction: A new excitation for CELP based speech coders

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ELECTRONICS LETTERS 1996年第6期32卷 520-521页

作者： Black, AW Kondoz, AM Evans, BG Centre for Satellite Engineering Research University of Surrey Guildford United Kingdom

The authors present a new secondary pulse excitation for linear prediction based analysis by synthesis speech coders. The structure of the excitation has been specifically designed to model characteristics in the speech waveform which the LTP memory fails to adequately represent. This is achieved using an excitation vector simply consisting of two pulses.

关键词： linear predictive coding speech coding

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Interframe Differential coding of Line Spectrum Frequencies

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第2期2卷 350-352页

作者： Erzin, Engin Cetin, A. Enis Bilkent Univ Dept Elect & Elect Engn TR-06533 Bilkent Turkey

Line spectrum frequencies (LSF's) uniquely represent the linear predictive coding (LPC) filter of a speech frame. In many vocoders LSF's are used to encode the LPC parameters. In this paper, an inter-frame differential coding scheme is presented for the LSF's. The LSF's of the current speech frame are predicted by using both the LSF's of the previous frame and some of the LSF's of the current frame. Then, the difference resulting from prediction is quantized.

关键词： Frequency linear predictive coding Testing Distortion measurement Nonlinear filters Speech coding Vocoders Poles and zeros Polynomials Signal processing

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Classification of cardiac arrhythmias using fuzzy ARTMAP

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IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1996年第4期43卷 425-430页

作者： Ham, FM Han, S Florida Institute of Technology Electrical Engineering West University Blvd Melbourne FL USA Electrical Engineering Florida Institute of Technology Melbourne FL USA

We have investigated the QRS complex, extracted from electrocardiogram (EGG) data, using fuzzy adaptive resonance theory mapping (ARTMAP) to classify cardiac arrhythmias. Two different conditions have been analyzed: normal and abnormal premature ventricular contraction (PVC), Based on MIT/BIH database annotations, cardiac beats for normal and abnormal QRS complexes were extracted from this database, scaled, and Hamming windowed, after bandpass filtering, to yield a sequence of 100 samples for each QRS segment, From each of these sequences, two linear predictive coding (LPC) coefficients were generated using Burg's maximum entropy method, The two LPC coefficients, along with the mean-square value of the QRS complex segment, were utilized as features for each condition to train and test a fuzzy ARTMAP neural network for classification of normal and abnormal PVC conditions, The test results show that the fuzzy ARTMAP neural network can classify cardiac arrhythmias with greater than 99% specificity and 97% sensitivity.

关键词： ART neural nets electrocardiography fuzzy neural nets linear predictive coding medical signal processing Burg's maximum entropy method Hamming window MIT/BIH database annotations abnormal QRS complexes abnormal premature ventricular contraction bandpass QRS complex fuzzy adaptive resonance theory map medical signal processing Fuzzy neural networks Hamming window Cardiac Arrhythmias Bandpass ART network

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Spectral compensation for linear prediction of speech signals in coloured noise

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ELECTRONICS LETTERS 1998年第11期34卷 1080-1081页

作者： Hu, HT Natl I Lan Inst Agr & Technol Dept Elect Engn I Lan 260 Taiwan

A method is presented to determine an appropriate portion that could be subtracted from the noise-contaminated autocorrelation functions before carrying out linear prediction analysis in coloured noise. This method guarantees the stability of the resulting linear prediction filter. The authors demonstrate the improvement using an objective measure based on a synthetic vowel.

关键词： linear prediction filter noise linear prediction spectral compensation speech recognition Codes speech processing synthetic vowel linear predictive coding Speech and audio signal processing noise-contaminated autocorrelation functions stability objective measure Speech recognition compensation coloured noise linear prediction analysis Information theory speech signals

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SPEAKER ADAPTIVE VECTOR QUANTIZATION OF LPC PARAMETERS OF SPEECH

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ELECTRONICS LETTERS 1988年第22期24卷 1392-1393页

作者： LEE, KY KONDOZ, AM EVANS, BG Surrey Univ. Guildford

Recently, it has been shown by Schroeder and Atal (see Proc. of ICASSP-85, p.937-40, 1985) that good quality speech at rates as low as 6 kbit/s can be achieved with CELP and its derivatives. However, in bringing down the bit rate even further these coding schemes have resorted to allocating fewer bits to the quantisation of the LPC parameters. It is known that CELP and its derivatives are sensitive to LPC parameters quantisation errors, and recently various schemes have been proposed to overcome this degradation. The authors describe how speaker adaptive vector quantisation (SAVQ) can be applied to the quantisation of LPC parameters and assess its performance when incorporated into a low bit rate coding scheme of 4.8 kbit/s

关键词： 4.8 kbit/s LPC parameters analogue-digital conversion code excited linear prediction digital communication systems digital transmission encoding filtering and prediction theory linear predictive coding low bit rate coding scheme speaker adaptive vector quantisation speech speech analysis and processing

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CONSTRAINED ITERATIVE SPEECH ENHANCEMENT WITH APPLICATION TO SPEECH RECOGNITION

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第4期39卷 795-805页

作者： HANSEN, JHL CLEMENTS, MA GEORGIA INST TECHNOL SCH ELECT ENGN ATLANTA GA 30332 USA

In this paper, an improved form of iterative speech enhancement for single channel inputs is formulated. The basis of the procedure is sequential maximum a posteriori estimation of the speech waveform and its all-pole parameters as originally formulated by Lim and Oppenheim, followed by imposition of constraints upon the sequence of speech spectra. The new approaches impose intraframe and interframe constaints on the input speech signal to ensure more speech-like formant trajectories, reduce frame-to-frame pole jitter, and effectively introduce a relaxation parameter to the iterative scheme. Recently discovered properties of the line spectral pair representation of speech allow for an efficient and direct procedure for application of many of the constraint requirements. Substantial improvement over the unconstrained method has been observed in a variety of domains. First, informal listener quality evaluation tests and objective speech quality measures demonstrate the technique's effectiveness for additive white Gaussian noise. A consistent terminating point for the iterative technique is also shown. Second, the algorithms have been generalized and successfully tested for noise which is nonwhite and slowly varying in characteristics. The current systems result in substantially improved speech quality and LPC parameter estimation in this context with only a minor increase in computational requirements. Third, the algorithms were evaluated with respect to improving automatic recognition of speech in the presence of additive noise, and shown to out-perform other enhancement methods in this application.

关键词： Speech enhancement Speech analysis linear predictive coding Maximum a posteriori estimation Testing Noise measurement Additive white noise Speech coding Parameter estimation Iterative algorithms

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LPC Spectral Moments for Clustering Acoustic Transients

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第3期1卷 362-368页

作者： Pinkowski, Ben Western Michigan Univ Dept Comp Sci Kalamazoo MI 49008 USA

Spectral moments (mean and coefficients of variation, skewness, and kurtosis) are assessed for 40 samples from 10 groups of acoustic transient signals differing in harmonic structure, duration, and degree of spectral overlap. Discri+minant analysis involving moments based on linear predictive coding (LPC) resulted in a higher recognition rate for pulsed-tone sounds (87%) that were more like human speech than for pure-tone sounds (70%). By contrast, classification based on moments calculated from the discrete Fourier transform (DFT) yielded 85% recognition for both groups. Cluster analyses indicated that LPC-based moments were more characteristic of relationships among the 10 sound groups and especially the 2 tonal groups, though results were somewhat dependent on LPC model order.

关键词： linear predictive coding Acoustic distortion Speech recognition Frequency Distortion measurement Speech analysis Humans Discrete Fourier transforms Speech processing Signal processing

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