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International Conference on Biomedical and Pharmaceutical Engineering

作者： Ng, Andrew Keong Koh, T. S. Baey, Eugene Puvanendran, K. Nanyang Technol Univ Sch Elect & Elect Engn Biomed & Pharmaceut Engn Cluster Singapore Singapore Nanyang Technol Univ Sch Elect & Elect Engn Ctr Modeling & Control Complex Syst Singapore Singapore Respiron Inc Murrysville PA USA Singapore Gen Hosp Sleep Disorders Unit Singapore Singapore

ISBN: (纸本)9789810579432

Correlation between snoring and Obstructive Sleep Apnea (OSA) has been studied over the past years using techniques such as sound intensity computation, power spectrum estimation, counting of snores, and more recently, pitch and jitter analysis. Although these studies have ascertained the relationship between snoring and OSA, they gave mixed outcomes in diagnosing OSA. In this paper, snore signals are analyzed using linear predictive coding (LPC), which is a powerful speech analytic tool. The noisy snore signals were first preprocessed using a Modified Normalized Least-Mean-Square (MNLMS) adaptive filter for noise cancellation, followed by the extraction of formant structures from the denoised snore signals to discriminate between benign and apneic snores. This initial study demonstrates that the formant structures of snore signals contain useful information for OSA detection, and suggests the feasibility of using snore signal analysis as a noninvasive and inexpensive diagnostic tool for mass screening of OSA.

关键词： obstructive sleep apnea snoring normalised least-mean-square adaptive filter linear predictive coding formant

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A sinusoidal LPC vocoder

A sinusoidal LPC vocoder

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7th IEEE Workshop on Speech coding

作者： Hedelin, P Chalmers Univ Technol Dept Signals & Syst Informat Theory Grp S-41296 Gothenburg Sweden

ISBN: (纸本)0780364163

Twenty years of work with sinusoidal modeling of speech has lead to very competitive principles of low rate coding. In this study, we discuss a few issues in the design of a sinusoidal coding system. We stress that by a careful design of all blocks of the encoder and decoder, allowing for some additional complexity, it is possible to build a low rate coder free of many of the artifacts associated with conventional vocoding systems. We focus this discussion on multi-band partial voicing and on parameter smoothing and interpolation.

关键词： Filter bank Filtering Interpolation linear predictive coding Power harmonic filters Signal generators Speech coding Speech enhancement Speech processing Vocoders

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All-pole modeling technique based on the weighted sum of the LSP polynomials

All-pole modeling technique based on the weighted sum of the...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Alku, P Bäckström, T Helsinki Univ Technol Lab Acoust & Audio Signal Proc FIN-02015 Helsinki Finland

ISBN: (纸本)0780374029

With the order of prediction equal to m, conventional linear prediction (LP) yields an all-pole filter that matches exactly the autocorrelation function of the input signal between indices 0 and m. This study presents a new technique, Weighted-Sum Line Spectrum Pair (WLSP), where an all-pole filter is defined by using a sum of weighted LSP polynomials. WLSP yields a stable all-pole filter of order m, whose autocorrelation function coincides to that of the input signal between indices 0 and m - 1. By sacrificing the exact matching of the autocorrelations at index m, WLSP models the autocorrelation of the input signal at the indices above m more accurately than conventional LP. The current paper presents mathematical properties of WLSP together with preliminary results on modeling of vowels. The results indicate that WLSP, in comparison to conventional LP, is able to yield all-pole filters that model especially the upper formants with a larger dynamic range between formant peaks and spectral valleys.

关键词： speech processing poles and zeros polynomials prediction theory spectral analysis speech coding linear predictive coding filtering theory all-pole filter weighted-sum line spectrum pair LSP polynomials linear prediction autocorrelation function vowels modeling upper formants formant peaks spectral valleys speech processing

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A new audio coder using a warped linear prediction model and the wavelet transform

A new audio coder using a warped linear prediction model and...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Ning, D Deriche, M Queensland Univ Technol Sch Elect Elect & Syst Engn Signal Proc Res Ctr Brisbane Qld 4001 Australia

ISBN: (纸本)0780374029

In this paper, we present results for a wavelet transform (WT) excited warped linear prediction (WLP) audio coder. In contrast to conventional LP, VVLP allows for the control of frequency resolution to closely match the response of the human auditory system. The structure of the system is similar to the transform-coded excitation techniques used in wideband speech coding, where LP has been replaced with WLP. Quantisation of the wavelet coefficients is aided by a psychoacoustic model to minimise the perceptually significant noise due to quantisation error. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent quality for a variety of speech and music signals at an average bit-rate of 64 kb/s. When compared to MPEG layer III at the same bit-rate, the coder delivers superior quality. The power of the proposed coder resides in its easy scalability to lower bitrates.

关键词： audio coding linear predictive coding transform coding wavelet transforms hearing quantisation (signal) speech coding signal sampling audio coding warped linear prediction wavelet transform frequency resolution human auditory system transform coding psychoacoustic model monophonic signals signal sampling music signals speech coding MPEG layer III scalability 44.1 kHz 64 kbit/s

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Subband synthesized LPC vector quantization (SBS-LPC-VQ)

Subband synthesized LPC vector quantization (SBS-LPC-VQ)

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7th IEEE Workshop on Speech coding

作者： Cao, BS Lucent Technol Bell Labs Innovat Wireless Network Syst Whippany NJ 07981 USA

ISBN: (纸本)0780364163

This paper presents a new LPC parameter quantization method - SubBand Synthesized LPC Vector Quantization (SBS-LPC-VQ) [9]. In the subband synthesis process, the relationships between subband spectra and whole band LPC spectrum are established and thus the vector quantized subband LPC parameters are mapped to the whole band LPC parameters. This new SBS-LPC-VQ method overcomes high complexity problem of vector quantization of LPC parameters and isolates the distortion within each subband during the VQ process. It also provides the flexibility of assigning the bits to be used for each subband, choosing the order for LPC filter and determining the number of bands for the subband classification. The Critical band Weighting Spectral Distortion Measure (CW-SDM), which is perceptually motivated objective measure by using a critical band weighting function, is used for measuring the distortion of quantized LPC spectrum. Using this kind of distortion measure, the experiments show that the SBS-LPC-VQ has obtained 24 bits/frame for coding whole 16th order LPC parameters with about 1 dB average spectral distortion. For comparison, the results by conventional Spectral Distortion Measure (SDM) an also presented in the paper.

关键词： Distortion measurement Filters Frequency linear predictive coding Network synthesis Signal analysis Signal processing Speech analysis Vector quantization Weight measurement

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Optimal Binary Measurement Matrices for Compressed Sensing

Optimal Binary Measurement Matrices for Compressed Sensing

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IEEE Information Theory Workshop (ITW)

作者： Tehrani, Arash Saber Dimakis, Alexandros G. Caire, Giuseppe Univ Southern Calif Dept Elect Engn Los Angeles CA 90089 USA Univ Texas Austin Dept Elect Engn Austin TX 78712 USA

ISBN: (纸本)9781479913213

We explicitly construct binary measurement matrices with good sparse approximation guarantees. Specifically, our measurement matrices have an order optimal number of measurements and have l(1)/l(1) approximation guarantee. Our construction uses the progressive edge growth technique. We apply coding theoretic results and rely on a recent connection of compressed sensing to LP relaxation for channel decoding.

关键词： channel coding compressed sensing linear predictive coding matrix algebra

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Multi stage matrix quantization for very low bit rate speech coding 3

Multi stage matrix quantization for very low bit rate speech...

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3rd IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC 01)

作者： Ozaydin, S Baykal, B Gazi Univ Dept Elect & Elect Engn Ankara Turkey

ISBN: (纸本)0780367200

Efficient quantization of linear predictive coding (LPC) filter coefficients play an essential role in very low bit rate speech coding systems. This paper examines a new suboptimal matrix quantization scheme for LPC parameters, called multi stage matrix quantization (MSMQ), which operates at bit rates between 400 and 800 bits/s, With the new matrix quantization method, using a 22,5 ms LPC analysis frame, spectral distortion about 1 dB is achieved at 800 bits/s, In the proposed coder, line spectral frequency (LSF) parameters of multiple consecutive frames are grouped into a superframe and jointly quantized. The new residual LSF vector quantization scheme gives a bit rate reduction in MSMQ without any additional complexity or storage. The new MSMQ leads into several schemes of various computational complexity/storage characteristics.

关键词： Bit rate Computational complexity Frequency Large scale integration linear predictive coding Nonlinear filters Spectral analysis Speech coding Vector quantization Wireless communication

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Isolated Word Recognition using Low Dimensional Features and Kernel based Classification

Isolated Word Recognition using Low Dimensional Features and...

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International Conference on Advances in Recent Technologies in Communication and Computing

作者： Nehe, N. S. Holambe, R. S. Pravara Rural Engn Coll Dept Instrumentat & Control Engg Loni Maharashtra India SGGS Inst Engn & Technol Dept Instrumentat Engg Nanded Maharashtra India

ISBN: (纸本)9781424451043

This paper describes polynomial kernel subspace approach to Isolated Word Recognition (IWR) systems. linear predictive coding (LPC) coefficients derived from wavelet sub-bands of speech frame were used as features. This approach represents mapping of speech features (input space) into a feature space via a non-linear mapping onto the principal components called Kernel linear Discriminant Analysis (KLDA). The non-linear mapping between the input space and the feature space is implicitly performed using the kernel-trick. This nonlinear mapping using KLDA increases the discrimination ability of a pattern classifier. The use of Wavelet sub-band based LPC features (WLPC) provide low dimensional features which reduce the memory requirement and KLDA provides the fast classification and recognition. Experimental results obtained on isolated word database show that the proposed technique is computationally efficient and performs well with less training data.

关键词： linear predictive coding Wavelet Transform Kernel linear Discriminant Analysis

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The Residual-Excited linear Prediction Vocoder with Transmission Rate Below 9.6 kbits/s

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IEEE Transactions on Communications 1975年第12期23卷 1466-1474页

作者： Un, Chong Kwan Magill, D. Thomas Telecommunications Sciences Center Stanford Research Institute Menlo Park Calif. 94025. United States

In this paper we present a new vocoder called the residual-excited linear prediction (RELP) vocoder. The concept of the RELP vocoder combines the advantages of linear predictive coding (LPC) and voice-excited vocoding. In the RELP system, vocal tract modeling is done by the LPC technique, and the LPC residual signal is used as the excitation signal. After lowpass filtering the residual signal is coded by adaptive delta modulation and is spectrally flattened before being fed in the LPC synthesizer. The range of the transmission rate is typically between 6 and 9.6 kbits/s;the synthetic speech in this range is quite good. As the transmission rate is lowered, the synthetic speech quality degrades very gradually. Since no pitch extraction is required, the vocoder is robust in any operating environment. Copyright © 1975 by The Institute of Electrical and Electronics Engineers, Inc.

关键词： Vocoders linear predictive coding Speech Adaptive filters Low pass filters Filtering Delta modulation Synthesizers Degradation Robustness

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Speech Recognition using Support Vector Machines 10

Speech Recognition using Support Vector Machines

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10th IEEE International Conference on Application of Information and Communication Technologies (AICT)

作者： Aida-zade, Kamil Xocayev, Anar Rustamov, Samir Azerbaijan Natl Acad Sci Inst Control Syst Baku Azerbaijan Qafqaz Univ Comp Engn Baku Azerbaijan ADA Univ Azerbaijan Natl Acad Sci Inst Control Syst Baku Azerbaijan

ISBN: (纸本)9781509018413

In this article we applied Support Vector Machines to acoustic model of Speech Recognition System based on MFCC and LPC features for Azerbaijani DataSet. This DataSet has been used for speech recognition by Multilayer Artificial Neural Network and achieved some results. The main goal of this work is applying SVM techniques to the Azerbaijan Speech Recognition System. The variety of results of SVM with different Kernel functions is analyzed in the training process. It is shown that SVM with radial basis and polynomial kernels give better recognition results that Multilayer Artificial Neural Network.

关键词： Speech Recognition Feature Extraction Support Vector Machines linear predictive coding Mel Frequency Cepstral Coefficients

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