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Could formant frequencies of snore signals be an alternative means for the diagnosis of obstructive sleep apnea?

SLEEP MEDICINE 2008年第8期9卷 894-898页

作者： Ng, Andrew Keong Koh, Tong San Baey, Eugene Lee, Teck Hock Abeyratne, Udantha Ranjith Puvanendran, Kathiravelu Nanyang Technol Univ Sch Elect & Elect Engn Singapore 639798 Singapore Respiron Inc Singapore 919191 Singapore Univ Queensland Sch Informat Technol & Elect Engn Brisbane Qld Australia Singapore Gen Hosp Sleep Disorders Unit Singapore 169608 Singapore

Objective: To study the feasibility of using acoustic signatures in snore signals for the diagnosis of obstructive sleep apnea (OSA). Methods: Snoring sounds of 30 apneic snorers (24 males;6 females: apnea-hypopnea index, AHI = 46.9 +/- 25.7 events/h) and 10 benign snorers (6 males;4 females;AHI = 4.6 +/- 3.4 events/h) were captured in a sleep laboratory. The recorded snore signals were preprocessed to remove noise, and subsequently, modeled using a linear predictive coding (LPC) technique. Formant frequencies (F1, F2, and F3) were extracted from the LPC spectrum for analysis. The accuracy of this approach was assessed using receiver operating characteristic curves and notched box plots. The relationship between AHI and F1 was further explored via regression analysis. Results: Quantitative differences in formant frequencies between apneic and benign snores are found in same- or both-gender snorers. Apneic snores exhibit higher formant frequencies than benign snores, especially F1, which can be related to the pathology of OSA. This study yields a sensitivity of 88%, a specificity of 82%, and a threshold value of F1 = 470 Hz that best differentiate apneic snorers from benign snorers (both gender combined). Conclusion: Acoustic signatures in snore signals carry information for OSA diagnosis, and snore-based analysis might potentially be a non-invasive and inexpensive diagnostic approach for mass screening of OSA. (c) 2007 Elsevier B.V. All rights reserved.

关键词： Obstructive sleep apnea Polysomnography Snoring Snore signals Acoustic analysis Formant frequencies linear predictive coding

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THE DISCRETE LYAPUNOV EQUATION IN CONTROLLABLE CANONICAL FORM

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IEEE TRANSACTIONS ON AUTOMATIC CONTROL 1981年第2期26卷 580-581页

作者： PTAK, V Institute of Mathematics Department of Functional Analysis Czechoslovak Academy of Science Prague Czech Republic

A new method of treating Lyapunov equations is proposed. If D and D^{-} are two self-adjoint one-dimensional matrices related in a certain way, then X-C^{\ast}XC=D if and only if X^{-1}-CX^{-1} C^{\ast}=D^{-} . As an application, a generalization of a recent result is given. If f is the vector f=(0,0...,0,1)^{T} , then the solution of X-CXC^{\ast} = ff^{\ast} is shown to be inverse of the Schur-Cohn matrix.

关键词： Equations Polynomials linear predictive coding Speech processing State estimation Parameter estimation Feature extraction

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KALMAN BACKWARD ADAPTIVE PREDICTOR COEFFICIENT IDENTIFICATION IN ADPCM WITH PCQ

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IEEE TRANSACTIONS ON COMMUNICATIONS 1980年第3期28卷 361-371页

作者： GIBSON, JD BERGLUND, VP SAUTER, LC TRW DEF & SPACE SYST GRP SATELLITE COMMUN SECTREDONDO BEACHCA 90278

Kalman backward adaptive predictor coefficient identification is combined with a modified pitch-compensating quantizer (MPCQ) to produce a high-performance adaptive differential pulse code modulation (ADPCM) system for operation at data rates of 12-16 kbits/s. The Kalman/MPCQ system is compared to an ADPCM system using a Kalman algorithm and robust Jayant qnantization and to a system with a fixed-tap predictor and MPCQ. The performance indicators are signal-to-quantization noise ratio (SNR), sound spectrogram analyses, and formal subjective listening tests. The SNR comparisons indicate that the Kalman/ MPCQ system has the highest SNR, followed by the fixed-tap/MPCQ system, and then the Kalman/robust Jayant system. Subjective listening test results show that the Kalman/MPCQ system is preferred over the fixed-tap/MPCQ system 100 percent of the time and over the Kalman/ robust Jayant system 80 percent of the time. Kalman adaptation thus provides an important perceptual effect not evident in the SNR's. The previously catastrophic effects of transmission errors on backward adaptive prediction are eliminated by simple ADPCM system modifications that do not affect the SNR or subjective quality of the output in the absence of errors for the five sentences studied. The problem of tandeming with a linear predictive coder (LPC) is investigated by using LPC processed speech as input to the three ADPCM systems and by using the output of the three ADPCM systems as input to an LPC analysis algorithm. For the LPC to ADPCM connection, the two systems with the MPCQ produce good quality output speech, while the system with robust Jayant quantization exhibits a fading phenomenon. For the ADPCM into LPC analysis, all three systems produce speech of approximately the same quality, with the fixedtap system being slightly, noisier. Using a distance measure proposed by Itakura, the predictor coefficients computed from the three ADPCM system outputs are compared with the predictor coefficien

关键词： Kalman filters linear predictive coding Signal to noise ratio Speech analysis Pulse modulation Noise robustness Acoustic noise Modulation coding Spectrogram Signal analysis

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RECURSIVE WINDOWING FOR GENERATING AUTO-CORRELATION COEFFICIENTS FOR LPC ANALYSIS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1981年第5期29卷 1062-1066页

作者： BARNWELL, TP School of Electrical Engineering Georgia Institute of Technology Atlanta GA USA

A method for recursively computing the autocorrelation estimates needed for LPC analysis in a vocoder environment has been developed theoretically and studied experimentally. The method has three specific advantages: 1) it requires very little memory for its implementation; 2) it is realized by a structure consisting of several identical modules; and 3) the effective window length may be changed without varying the structure. Experimental results showed the speech quality to be comparable to and slightly better than that produced by an auto-correlation LPC vocoder using a Hamming window.

关键词： Autocorrelation linear predictive coding Speech Vocoders Buffer storage Recursive estimation Quantization Feature extraction Digital filters Logic

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AN ALGORITHM FOR DIRECT COMPUTATION OF 2-D linear PREDICTION COEFFICIENTS

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1993年第2期41卷 996-1000页

作者： KUDUVALLI, GR RANGAYYAN, RM Department of Electrical and Computer Engineering University of Calgary Calgary AB Canada

We present some simplifications to the method of computing two-dimensional (2-D) linear prediction (LP) coefficients (LPC's) directly from the image data using an extension of the multichannel version of the Burg algorithm presented by Therrien and El-Shaer, This simplification results from forcing the structure of the 2-D autocorrelation matrices on the multichannel version of the Burg algorithm. In addition to computing the 2-D LPC's, the method can be used for the computation of prediction errors directly from the data. Results of application of the method to 2-D LP coding of images and 2-D autoregressive (AR) spectral estimation are presented.

关键词： Two dimensional displays linear predictive coding Image coding

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AN INTERPRETATION OF THE LOG LIKELIHOOD RATIO AS A MEASURE OF WAVEFORM CODER PERFORMANCE

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1980年第3期28卷 318-323页

作者： CROCHIERE, RE TRIBOLET, JM RABINER, LR Univ Tecn Lisboa INST SUPER TECN LISBON 1 PORTUGAL

The log likelihood measure has been widely used in speech research for comparing speech signals. Recently, it has been proposed as a measure for assessing the quality of coded speech. In this paper we present an interpretation of the log likelihood ratio measure within the theoretical framework of a waveform coder distortion model. We then discuss the implications of this interpretation and show how it can be applied to the formulation of better objective measures of waveform coder performance.

关键词： Distortion measurement Acoustic distortion linear predictive coding Acoustic measurements Filters Speech analysis Length measurement Design optimization Speech processing Noise measurement

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EFFECT OF CHANNEL ERRORS ON THE PERFORMANCE OF LPC VOCODERS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1983年第1期31卷 234-237页

作者： UN, CK LEE, SJ AGCY DEF DEV DAEJONSOUTH KOREA

The effect of channel errors on the performance of LPC vocoders has been investigated. The vocoder performance is fairly robust when the error rate is 10 -3 or less. Among the various parameters, the first two reflection coefficients and pitch are most sensitive to channel errors. When these parameters are coded with an error protection scheme, the performance is robust even when the error rate is as high as 10 -2 .

关键词： linear predictive coding Vocoders Working environment noise Error analysis Reflection Degradation Interpolation Solids Speech processing Protection

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Comments on "Reducing Computation in HMM Evaluation"

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第4期2卷 542-543页

作者： Mitchell, Carl Harper, Mary Jamieson, Leah Purdue Univ Sch Elect Engn W Lafayette IN 47907 USA

We show that HMM word recognition using Deller and Snider's "any path" procedure makes an assumption of independence that is not made by either the forward or Viterbi algorithms. We also point out that a... 详细信息

关键词： Hidden Markov models Speech coding linear predictive coding Bit rate Viterbi algorithm Additive noise Background noise Frequency dependence Signal resolution Signal processing

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Several Classifiers for Intruder Detection Applications 9

Several Classifiers for Intruder Detection Applications

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International Conference on Speech Technology and Human-Computer Dialogue (SpeD)

作者： Buhus, Elena Roxana Grama, Lacrimioara Rusu, Corneliu Tech Univ Cluj Napoca Fac Elect Telecommun & Informat Technol Basis Elect Dept Signal Proc Grp Cluj Napoca Romania

ISBN: (纸本)9781509064977

The goal of this work is to present some possible intruder detection systems and the influence of impulse-like signals upon the overall classification accuracy. Two different scenarios are used: in the first scenario five sound classes are considered (last class belong to impulsive sounds gunshots), while in the second scenario we dropped out the impulsive sound class. More classifiers are used in both scenarios and different number of features are considered. An improvement in the classification accuracy is obtained within the second scenario. The highest accuracy for the first scenario is for J48 classifier using 51 features, while for the second scenario the highest accuracy is attained for Simple Logistic classifier wit 101 features.

关键词： audio signal classification intruder detection linear predictive coding WEKA

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A PULSE DRIVING FUNCTION GENERATOR FOR LPC SYNTHESIS OF VOICED SEGMENTS OF SPEECH

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1981年第6期29卷 1113-1116页

作者： HARTWELL, WT PREZAS, DP Bell Laboratories Inc. Naperville IL USA

Most of the pitch detection schemes for voiced segments of speech measure the pitch period as an integer multiple of the sampling period. Averaging techniques on more than one frame, for smoothing purposes, will result in pitch periods which are noninteger multiples of the sampling period. The same result is expected when interpolation takes place by changing the pitch several times within the same frame. Finally, a similar effect is observed, even without smoothing or interpolation, in those pitch detection schemes that use averaging of distances within the same frame to calculate the final pitch period for the frame. The algorithm described here permits the synthesis of male, female, and children's voices using noninteger pitches. The impact of this technique on current LPC implementations is marginal at low pitch frequencies while, at higher ones, the improvement is detectable. It is believed that as other sources of error in LPC systems are reduced, this technique will prove to be useful in improving synthetic speech quality. This paper deals with two items. It discusses the effects of truncation and rounding of noninteger pitch periods and describes the design of a pulse driving function generator, whose pitch period is a noninteger multiple of a fixed sampling period.

关键词： Speech synthesis Signal generators linear predictive coding Sampling methods Pulse shaping methods Control system synthesis Shape Pulse measurements Interpolation Smoothing methods

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