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An efficient approach using HOS-based parameters in the LPC residual domain to classify breathy and rough voices

BIOMEDICAL SIGNAL PROCESSING AND CONTROL 2011年第2期6卷 186-196页

作者： Lee, JiYeoun Jeong, Sangbae Hahn, Minsoo Sprecher, Alicia J. Jiang, Jack J. Univ Wisconsin Sch Med Med Sci Ctr 5745A Div Otolaryngol Head & Neck Surg Madison WI 53706 USA Gyeongsang Natl Univ Sch Elect & Elect Engn Div Elect Engn Jinju 660701 South Korea Korea Adv Inst Sci & Technol Sch Engn Div Multimedia Commun & Proc Taejon 305701 South Korea

Although a considerable number of studies have been focused on the analysis of pathological voices using conventional parameters such as jitter, shimmer, and signal-to-noise ratio (SNR), these parameters have been found to be sensitive to variations in pitch extraction algorithm and cannot analyze severely disordered voice signals which exhibit irregular or aperiodic waveforms. In this paper, higher-order statistics (HOSs) analysis, which is independent of pitch period, is derived from linear predictive coding (LPC) residuals to describe breathy and rough voices. Recordings of a sustained /a/ from 23 individuals with breathy voices and 30 individuals with rough voices were collected from the disordered voice database distributed by the Japanese Society of Logopedics and Phoniatrics. We extracted conventional parameters as well as the HOS-based parameters such as the normalized skewness and the normalized kurtosis. On the other hand, we calculated HOS-based parameters from the LPC residual domain. The results showed that the HOS-based parameters and the HOS-based parameters estimated from the LPC residual are different for rough and breathy voices. Conventional parameters were not distinctive for these voices. Classification and regression tree (CART) was used to combine multiple parameters and to classify breathy and rough voices. Using the HOS-based parameters, the CART achieved an accuracy of 85.0% with the optimal decision tree generated by means of the normalized skewness and kurtosis. When the HOS-based parameters using LPC residual were used, the optimal decision tree was 88.7% accurate and the variances of the normalized skewness and kurtosis were included. (C) 2010 Elsevier Ltd. All rights reserved.

关键词： Pathological voice quality Higher-order statistics linear predictive coding Classification and regression tree

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ON THE USE OF INSTANTANEOUS AND TRANSITIONAL SPECTRAL INFORMATION IN SPEAKER RECOGNITION

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1988年第6期36卷 871-879页

作者： SOONG, FK ROSENBERG, AE AT and T Bell Laboratories Inc. Murray Hill NJ USA

The use of instantaneous and transitional spectral representations of spoken utterances for speaker recognition is investigated. linear-predictive-coding (LPC)-derived cepstral coefficients are used to represent instantaneous spectral information, and best linear fits of each cepstral coefficient over a specified time window are used to represent transitional information. An evaluation has been carried out using a database of isolated digit utterances over dialed-up telephone lines by 10 talkers. Two vector quantization (VQ) codebooks, instantaneous and transitional, were constructed from each speaker's training utterances. The experimental results show that the instantaneous and transitional representations are relatively uncorrelated, thus providing complementary information for speaker recognition. A rectangular window of approximately 100 ms duration provides an effective estimate of the transitional spectral features for speaker recognition. Also, simple transmission channel variations are shown to affect both the instantaneous spectral representations and the corresponding recognition performance significantly, while the transitional representations and performance are relatively resistant.< >

关键词： Speaker recognition Cepstral analysis linear predictive coding Spectral analysis Databases Data mining Speech analysis Filter bank Distortion measurement Telephony

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Lossless Pole-Zero Modeling of Speech Signals

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第3期1卷 269-276页

作者： Lim, Il-Taek Lee, Byeong Gi Seoul Natl Univ Dept Elect Engn Seoul 151742 South Korea

In this paper we established a theory on lossless pole-zero modeling of speech signals for the description of nasal sounds. The theory is based on a generalized vocal tract tube model which consists of the main vocal tract, the oral tract, and the nasal tract. We first derived a new pole-zero type transfer function, which turned out to be a generalized version of the existing all-pole type transfer function. We then investigated fundamental properties of the generalized vocal tract model employing the concept of discrete-time reactance. Based on those properties, we outlined a procedure to evaluate the reflection coefficients for the model. The assumption of losslessness in the modeling led us to the following two important properties. First, the combination of two lattice structures representing the oral and the nasal tracts form one larger lattice structure when viewed at their joint point called the branch boundary, and secondly, the oral and the nasal tract respectively render discrete-time reactances G(C)(z) and G(D)(z), whose convex combination generates the discrete-time reactance G(A)(z) at the branch boundary. The first property enabled us to compute G(A)(z), and the second property helped us separate G(A)(z) into G(C)(z) and G(D)(z).

关键词： Transfer functions linear predictive coding Speech analysis Filters Nonlinear equations Lattices Reflection Signal synthesis Speech synthesis

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An 8-kb/s conjugate structure CELP (CS-CELP) speech coder

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第6期4卷 401-411页

作者： Kataoka, A Moriya, T Hayashi, S NTT Human Interface Laboratories Musashino Tokyo Japan

This paper describes a high-quality 8-kb/s speech coder called conjugate structure code-excited linear prediction (CS-CELP) with 10-ms frame length, To provide a short delay and high quality under both error-free add channel error conditions, it uses three dew schemes: line spectrum pair (LSP) quantization using interframe prediction, preselection in the codebook search, and gain vector quantization (VQ) with backward prediction;LSP parameters are quantized by using multistage VQ with moving-average (MA) prediction, This scheme can operate efficiently with various frequency responses of speech. The preselection of the codebook reduces computational complexity and improves robustness to channel errors. The gain VQ with backward prediction can provide high quality and robustness without transmission of input speech power information, A conjugate structure for both random codebook and gain codebook is introduced to improve the ability to handle random bit errors and to reduce codebook storage memory requirements, Subjective testing indicates that the quality of this coder is equivalent to that of 32-kb/s adaptive differential pulse code modulation (ADPCM) under error-free conditions, Testing has further demonstrated that the coder is robust against random bit errors.

关键词： Speech coding linear predictive coding Speech analysis Robustness Testing Pulse modulation North America Delay Personal communication networks Decoding

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Simulation model of the vocal tract filter for speech synthesis

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SIMULATION 1996年第4期67卷 241-246页

作者： AlAkaidi, MM [o_1]Department of Electronic Engineering DeMontfort University Leicester United Kingdom

This paper addresses the techniques used to produce a relatively low bit rate for speech coding. The technique initially explored in this work is the linear predictive coder, the most popular method referenced to date. This technique uses algorthims that describe the speech production process during voice and unvoiced sounds. The linear predictive coder attempts to approximate the vocal tract filter over a short period of time. The model, coupled with a specific excitation, can be used for speech synthesis.

关键词： speech linear predictive coding synthesis autocorrelation

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Robust speaker identification via fusion of subglottal resonances and cepstral features

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2017年第4期141卷 EL420-EL426页

作者： Guo, Jinxi Yang, Ruochen Arsikere, Harish Alwan, Abeer Univ Calif Los Angeles Dept Elect Engn Los Angeles CA 90095 USA

This letter investigates the use of subglottal resonances (SGRs) for noise-robust speaker identification (SID). It is motivated by the speaker specificity and stationarity of subglottal acoustics, and the development of noise-robust SGR estimation algorithms which are reliable at low signal-to-noise ratios for large datasets. A two-stage framework is proposed which combines the SGRs with different cepstral features. The cepstral features are used in the first stage to reduce the number of target speakers for a test utterance, and then SGRs are used as complementary second-stage features to conduct identification. Experiments with the TIMIT and NIST 2008 databases show that SGRs, when used in conjunction with power-normalized cepstral coefficients and linear prediction cepstral coefficients, can improve the performance significantly (2%-6% absolute accuracy improvement) across all noise conditions in mismatched situations. (C) 2017 Acoustical Society of America

关键词： AUTOMATIC speech recognition RESONANCE CEPSTRUM analysis (Mechanics) linear predictive coding NOISE

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ADAPTIVE BLOCK TRANSFORM coding OF SPEECH BASED ON LPC VECTOR QUANTIZATION

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第12期39卷 2611-2620页

作者： HUSSAIN, Y FARVARDIN, N UNIV MARYLAND SYST RES CTR COLLEGE PK MD 20742 USA

In this paper we describe several adaptive block transform speech coding systems based on vector quantization of LPC parameters. In order to account for the power fluctuations, the speech signal is normalized to have a unit-energy prediction residual. The temporal variations in the short-term spectrum, on the other hand, are taken into account by vector quantizing the LPC parameters associated with the vector of speech samples and transmitting the codeword index. Also, a variation of the scheme in which the pitch information is used to better estimate the spectrum is considered. For each block, based on the codevector associated with the input vector, an optimum bit assignment map is used to quantize the transform coefficients. We consider two types of zero-memory quantizers for encoding the transform coefficients, namely the Lloyd-Max quantizer and the entropy-coded quantizer. The performance of these schemes is compared with other adaptive transform coding schemes. We show by means of simulations that the system based on entropy-coded quantizer design leads to very high performance and in most cases as much as 5-dB performance improvement in terms of segmental signal-to-noise ratio is observed over the adaptive block transform coding scheme of Zelinski and Noll. The effect of incorporating the pitch information in the coder is studied and numerical results are presented. The effects of the bit rate and the size of the codebook on the performance of the systems are also studied in detail.

关键词： Transform coding Speech coding linear predictive coding Vector quantization Bit rate Hidden Markov models State estimation Encoding Signal design Signal to noise ratio

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A MULTIRATE VOICE DIGITIZER BASED UPON VECTOR QUANTIZATION

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IEEE TRANSACTIONS ON COMMUNICATIONS 1982年第4期30卷 721-727页

作者： REBOLLEDO, G GRAY, RM BURG, JP TIME & SPACE PROC INC SANTA CLARACA 95051 STANFORD UNIV INFORMAT SYST LABSTANFORDCA 94305

The importance of integrating voice and data over digital networks has increased during the last few years primarily because of the growing popularity of such networks. Of particular interest are efficient voice digitizing terminals, capable of operating at various data rates in both circuit-switched and packet-switched data networks. Several such terminals, including two or more speech compression algorithms, have been proposed and implemented. Typically the terminal switches between a low-rate (500 - 4000 bits/s) vocoding scheme and a medium-rate (7000 - 16000 bits/s) waveform coding algorithm, depending on, among other things, the network congestion and on the desired voice quality and robustness. We here describe the design and simulation of a multirate voice digitizer (MRVD) that switches between two speech compression systems, each based on a recently developed vector quantization (VQ) coding technique. This technique consists of the off-line interactive design of a codebook minimizing an average distortion measure, followed by the use of the codebook in an on-line nearest neighbor encoding scheme. One of the two systems is a rate-distortion speech coder that resembles a linear predictive coding (LPC) speech compression system but has a much lower rate (800 bits/s and below). We call this the LPC-VQ system, and it is similar to other previously reported systems [15],[19],[21]. The only difference is that the LPC parameters are extracted using the Burg method instead of the autocorrelation method. We here show that this provides both qualitative and quantitative improvements. The other system of our MRVD is a residual-excited linear predictive (RELP) speech compression system using VQ in both model selection and residual digitization. The residual waveform is digitized at 1 or 2 bits/sample, resulting in rates of 7300 and 13800 bits/s, respectively. We call this the RELP-VQ system. When compared to other RELP systems [6]-[8], it is shown to have a simpler archi

关键词： Vector quantization Speech coding linear predictive coding Switches Circuits Compression algorithms Robustness Distortion measurement Nearest neighbor searches Encoding

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Fixed-Lag Smoothing for Low-Delay predictive coding with Noise Shaping for Lossy Networks

Fixed-Lag Smoothing for Low-Delay Predictive Coding with Noi...

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Data Compression Conference (DCC)

作者： Thomas Arildsen Jan Ostergaard Manohar N. Murthi Soren Vang Andersen Soren Holdt Jensen Department of Electronic Systems University of Aalborg Denmark Electrical and Computer Engineering University of Miami FL USA

ISBN: (纸本)9781424464258;9780769539942

We consider linear predictive coding and noise shaping for coding and transmission of auto-regressive (AR) sources over lossy networks. We generalize an existing framework to arbitrary filter orders and propose use of fixed-lag smoothing at the decoder, in order to further reduce the impact of transmission failures. We show that fixed-lag smoothing up to a certain delay can be obtained without additional computational complexity by exploiting the state-space structure. We prove that the proposed smoothing strategy strictly improves performance under quite general conditions. Finally, we provide simulations on AR sources, and channels with correlated losses, and show that substantial improvements are possible.

关键词： Smoothing methods predictive coding Noise shaping linear predictive coding Propagation losses Filters Decoding Delay Computational complexity Computational modeling

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predictive SVD-transform coding of speech with adaptive vector quantization

Predictive SVD-transform coding of speech with adaptive vect...

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

作者： E. Masgrau J.A. Rodriguez-Fonollosa J.R. Mallafre Department T.S.C. E.T.S.I. Telecomunicacion UPC Barcelona Spain

A predictive transform coder using singular value decomposition (SVD) and adaptive vector quantization (AVQ) is presented. The linear predictive coding (LPC) excitation is obtained by a weighting VQ quantization of the SVD transforms of the prediction residual. The orthogonal representation of the excitation provided by the SVD transform permits a fine quantization of the perceptually important components. Some dynamic bit allocation approaches based on the relative importance of the singular values are described. The integration of the usual perceptual noise-shaping is easy, and a simple noise-shaping based on the direct reduction of the singular values spread is presented. The LPC coefficients are quantized by an adaptive multistage VQ approach. The results show good performance within the range of 7-8 kb/s.< >

关键词： Speech coding Vector quantization linear predictive coding Filters Bit rate Noise shaping Signal processing algorithms Array signal processing Telecommunications Product codes

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