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Iterative algorithm for design of vector quantisation codebooks in nonorthogonal domains

ELECTRONICS LETTERS 2002年第12期38卷 607-609页

作者： Mikhael, WB Krishnan, V Univ Cent Florida Coll Engn & Comp Sci Dept Elect Engn Orlando FL 32816 USA

An iterative algorithm for codebook accuracy enhancement, applicable to both waveform and linear prediction (LP) model-based vector quantisation in nonorthogonal domains, is developed and presented. Sample results are... 详细信息

关键词： transform coding adaptive codebook accuracy enhancement linear predictive coding iterative methods iterative algorithm vector quantisation Codes Interpolation and function approximation (numerical analysis) nonorthogonal domains linear prediction model-based vector quantisation codebook design waveform vector quantisation adaptive codes

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Natural quality two-band LPC coding of speech at 880 bit/s with frame interpolation

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ELECTRONICS LETTERS 2002年第6期38卷 292-294页

作者： Han, WJ Kim, EK Oh, YH Korea Adv Inst Sci & Technol Dept Elect Engn & Comp Sci Div Comp Sci Yusong Gu Taejon 305701 South Korea

A novel frame interpolation technique for two-band linear predictive coding (LPC) vocoders is proposed for maintaining natural speech quality at bit rates below I kbit/s. Experimental results show that the speech qual... 详细信息

关键词： natural speech quality interpolation linear predictive coding 880 bit/s speech coding vocoders frame interpolation two-band linear predictive coding vocoder Speech and audio coding bit rate Codecs, coders and decoders

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Digital acoustic analysis of five vowels in maxillectomy patients

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JOURNAL OF ORAL REHABILITATION 2002年第7期29卷 649-656页

作者： Sumita, YI Ozawa, S Mukohyama, H Ueno, T Ohyama, T Taniguchi, H Tokyo Med & Dent Univ Dept Maxillofacial Prosthet Grad Sch Bunkyo Ku Tokyo 1138549 Japan Tokyo Med & Dent Univ Dept Sports Med Dent Grad Sch Tokyo 1138549 Japan Tokyo Med & Dent Univ Dept Removable Prosthodont Grad Sch Tokyo 1138549 Japan

The aim of the study was to characterize the acoustics of vowel articulation in maxillectomy patients. Digital acoustic analysis of five vowels, /a/, /e/, /i/, /o/ and /u/, was performed on 12 male maxillectomy patients and 12 normal male individuals. A simple set of acoustic descriptions called the first and second formant frequencies, F1 and F2, were employed and calculated based on linear predictive coding. The maxillectomy patients had a significantly lower F2 for all five vowels and a significantly higher F1 for only /i/ vowel. From the data plotted on an F1-F2 plane in each subject, we determined the F1 range and the F2 range, which are the differences between the minimum and the maximum frequencies among the five vowels. The maxillectomy patients had a significantly narrower F2 range than the normal controls. In contrast, there was no significant difference in the F1 range. These results suggest that the maxillectomy patients had difficulty in controlling F2 properly. In addition, the speech intelligibility (SI) test was performed to verify the results of this new frequency range method. A high correlation between the F2 range and the score of SI test was demonstrated, suggesting that the F2 range is effective in evaluating the speech ability of maxillectomy patients.

关键词： vowel maxillectomy digital acoustic analysis linear predictive coding formant

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Discriminative training for neural predictive coding applied to speech features extraction

Discriminative training for neural predictive coding applied...

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International Joint Conference on Neural Networks (IJCNN)

作者： M. Chetouani B. Gas J.L. Zarader C. Chavy Laboratoire des Instruments et Systèmes (d'IIe de France (LISIF) Université Paris VI Paris Cedex 05 FRANCE

We present a predictive neural network called neural predictive coding (NPC). This model is used for nonlinear discriminant features extraction applied to phoneme recognition. We validate the nonlinear prediction improvement of the NPC model. We also, present a new extension of the NPC model: NPC-3. In order to evaluate the performances of the NPC-3 model, we carried out a study of Darpa-Timit phonemes (in particular /b/, /d/, /g/ and /p/, /t/, /q/ phonemes) recognition. Comparisons with traditional coding methods are presented. We also show how an adaptative constraint allows improvements on the recognition task.

关键词： predictive coding Speech coding Feature extraction predictive models Nonlinear filters linear predictive coding Neural networks Speech recognition Mel frequency cepstral coefficient Instruments

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The Modular Neural predictive coding architecture

The Modular Neural Predictive Coding architecture

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International Conference on Neural Information Processing

作者： M. Chetouani B. Gas J.L. Zarader Laboratoire des Instruments et Systemes d'Ile de France (LISIF) Universite Paris VI Paris France

ISBN: (纸本)9810475241

We present a new architecture called the Modular Neural predictive coding architecture (Modular NPC). This architecture is used for speech discriminant feature extraction (DFE). We present an application of the modular NPC architecture on phoneme recognition task. The phonemes which are extracted from the Darpa-Timit speech database are: vowels, /b/-/d/-/g/ and /p/-/t/-/k/ phonemes. Comparisons with coding methods (LPC, MFCC, PLP) are presented.

关键词： predictive coding Feature extraction linear predictive coding Speech recognition predictive models Mel frequency cepstral coefficient linearity Optimized production technology Speech processing Vectors

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Scalable innovation coding

Scalable innovation coding

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Nordén, F Hedelin, P Chalmers Univ Technol Informat Theory Lab SE-41296 Gothenburg Sweden

ISBN: (纸本)0780374029

Low-complexity scalable methods are of importance to achieve multi-rate and variable rate speech coding. Adopting the CELP concept, we focus on the innovation coding, suggesting an adaptive companding VQ. We present a scheme having essentially no increase in complexity as the rate is increased, yet having a competitive distortion performance. This is acheived by avoiding an explicit perceptual filtering step in the coding, still utilizing close to optimal VQ techniques in a perceptual domain. Subjective and objective distortion performance is better than, or in parity with, that of conventional white noise excitation methods or multipulse structures.

关键词： linear predictive coding speech coding vector quantisation compandors scalable innovation coding multi-rate coding variable rate coding speech coding CELP adaptive companding VQ perceptual domain objective distortion performance subjective distortion performance

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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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Packet loss concealment based on sinusoidal extrapolation

Packet loss concealment based on sinusoidal extrapolation

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Lindblom, J Hedelin, P Chalmers Univ Technol Informat Theory Lab SE-41296 Gothenburg Sweden

ISBN: (纸本)0780374029

In this paper, a packet loss concealment scheme based on sinusoidal extrapolation is proposed. It is to be implemented at the receiver end of a packetized speech transmission system and conceals frame-erasures in a PCM waveform. The scheme operates on the source-filter components of the speech signal, which are obtained through linear Prediction (LP) analysis. When one or more speech frames are missing, the LP residual and filter coefficients are extrapolated separately. The residual is extrapolated based on sinusoidal modeling of the last correct residual frame, whereas the latest filter is repeated and bandwidth expanded. The proposed scheme is compared to two benchmark systems by means of subjective testing. The benchmark systems are based on repetitions of the last pitch period;one operates on the source-filter components, the other directly on the speech signal. It is noted that the systems operating on the source-filter components are preferred by the listeners, and that the proposed system is clearly superior to both benchmark systems.

关键词： speech coding extrapolation pulse code modulation linear predictive coding filtering theory voice communication packet loss concealment sinusoidal extrapolation packetized speech transmission system frame-erasures concealment PCM waveform source-filter components linear prediction analysis filter coefficients LP residual subjective testing last pitch period repetitions source filter components packet voice communication

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Joint optimization of model and excitation in parametric speech coders

Joint optimization of model and excitation in parametric spe...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Lashkari, K Miki, T DoCoMo USA Labs Inc San Jose CA 95110 USA

ISBN: (纸本)0780374029

This paper presents a new Analysis-by-Synthesis (AbS) technique for joint optimization of the excitation and model parameters based on minimizing the closed loop synthesis error instead of the linear prediction error. By minimizing the synthesis error, the analysis and synthesis stages become more compatible. Using a gradient search in the root domain, model parameters for a given excitation are optimized to minimize the error between the original and the synthesized speech. Since the optimization starts from the LPC solution, the synthesis error is guaranteed to be lower than that obtained using the LPC coefficients. For multipulse LPC, there is a 0.5-1 dB improvement in the segmental SNR for male and female speakers over 4 to 6 second long sentences. Listening tests and objective MOS scores confirm the improved speech quality. By adding an extra optimization step, the technique can be incorporated into the LPC, multi-pulse LPC and CELP-type speech coders.

关键词： speech coding vocoders linear predictive coding optimisation speech enhancement speech synthesis minimisation gradient methods search problems joint optimization parametric speech coders analysis-by-synthesis technique excitation parameters model parameters closed loop synthesis error minimization gradient search multipulse LPC segmental SNR listening tests objective MOS scores improved speech quality CELP-type speech coders

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