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Source-system windowing for speech analysis and synthesis

IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第2期4卷 133-137页

作者： Yegnanarayana, B Murthy, PS Indian Institute of Technology Madras Chennai Tamil Nadu IN Dept. of Comput. Sci. & Eng. Indian Inst. of Technol. Madras India

In this correspondence, a new method of analysis of speech is proposed that will bring out variations in vocal tract system characteristics in short (2-4 ms) segments. In this method, the source and system components ... 详细信息

关键词： Speech analysis Speech synthesis Production systems Signal synthesis Autocorrelation Signal generators Time varying systems Continuous production linear predictive coding Independent component analysis

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LPC analysis incorporating spectral interpolation for speech coding

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ELECTRONICS LETTERS 1999年第3期35卷 200-201页

作者： Lee, MS Kim, HK Lee, HS Korea Adv Inst Sci & Technol Dongdaemun Gu Seoul 130012 South Korea SK Telekom Cent R&D Ctr Yusong Gu Taejon 305348 South Korea

A linear predictive coding (LPC) analysis scheme which is applicable to speech coding is proposed. The analysis method, called interpolative LPC (ILPC) analysis, estimates the spectral envelope by incorporating the interpolation characteristics into the LPC analysis. The ILPC analysis reduces average spectral distortion and the percentage of outlier frames, compared with the conventional LPC analysis followed by linear interpolation.

关键词： LPC analysis Speech processing techniques interpolation linear predictive coding speech coding spectral envelope estimation Speech and audio coding Interpolation and function approximation (numerical analysis) Signal processing theory spectral interpolation interpolative LPC

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Unequal error protection and source-channel decoding of CELP speech

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ELECTRONICS LETTERS 2002年第7期38卷 347-349页

作者： Nazer, A Alajaji, F Queens Univ Dept Math & Stat Kingston ON K7L 3N6 Canada

The reliable communication of FS CELP 10 16 encoded speech over very noisy channels is investigated. Using second-order Markov chains it is shown that over one-quarter of the CELP bits in every frame of speech are redundant. An unequal error protection coding scheme, which exploits this residual redundancy, is proposed for sending the CELP parameters over Gaussian and Rayleigh fading channels. Simulations indicate substantial coding gains over conventional systems.

关键词： speech coding Gaussian channels combined source-channel coding reliable communication coding gains error correction codes voice communication decoding CELP speech second-order Markov chains linear predictive coding source-channel decoding Rayleigh channels telecommunication network reliability Markov processes unequal error protection coding FS CELP 1016 encoded speech redundancy residual redundancy very noisy channels Speech and audio coding Rayleigh fading channels

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Improving pitch estimation for efficient multiband excitation coding of speech

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ELECTRONICS LETTERS 1996年第10期32卷 870-872页

作者： Chan, CF Yu, EWM Dept. of Electron. Eng. City Polytech. of Hong Kong Kowloon

The authors describe an improved pitch detection algorithm for efficient multiband excitation (MBE) coding of speech. The improved algorithm adds a corrective measure to the error measure for spectrum matching employed in conventional MBE pitch analysis. This corrective measure effectively applies equal weighting Lo all harmonic bands by normalising the error energy in each band. The result is the reduction of gross pitch errors owing to pitch doublings,An additional advantage of using this corrective measure is that, because this measure is based on a sum-of-product formula, comparisons of matching scores may be performed with partial-sums computed during the evaluation of the measure, therefore, facilitating fast searching of the optimum pitch period, Simulation results show that, with the deployment of the improved algorithm, pitch tracking procedure is no longer needed;and the coding delay of the MBE coder;is significantly reduced.

关键词： speech coding linear predictive coding

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Nonlinear Long-Term Prediction of Speech Based on Truncated Volterra Series

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IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2012年第3期20卷 1069-1073页

作者： Despotovic, Vladimir Goertz, Norbert Peric, Zoran Univ Belgrade Tech Fac Bor Bor 19210 Serbia Vienna Univ Technol Inst Telecommun A-1040 Vienna Austria Univ Nis Fac Elect Engn Nish 18000 Serbia

Previous studies of nonlinear prediction of speech have been mostly focused on short-term prediction. This paper presents long-term nonlinear prediction based on second-order Volterra filters. It will be shown that the presented predictor can outperform conventional linear prediction techniques in terms of prediction gain and "whiter" residuals.

关键词： linear predictive coding long-term prediction nonlinear filters volterra series

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Subspectral modeling in filter banks

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1995年第12期43卷 3050-3053页

作者： Benyassine, A Akansu, AN Department of Electrical and Computer Engineering Center for Communications and Signal Processing Research New Jersey Institute of Technology University Heights Newark NJ USA

This correspondence deal with spectral modeling in filter banks. It is shown, both theoretically and experimentally, that subspectral modeling is superior to full spectrum modeling if performed before the rate change. The price paid for this performance improvement is an increase of computations. A few different signal sources were considered in this study. It is shown that the performance of AR and ARMA techniques are comparable in subspectral modeling. The first is desired because of its simplicity. As an application of this study, we implemented a CELP based speech codec embedded in a filter bank structure. We found that there were no performance improvements of subband CELP technique over the fullband case. The theoretical reasonings of the experimental results are also given in this correspondence.

关键词： Channel bank filters linear predictive coding Filter bank Signal resolution Spectral analysis predictive models Speech coding Transfer functions Speech codecs Bandwidth

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The MDL criterion for rank determination via effective singular values

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1998年第6期46卷 1741-1744页

作者： Zarowski, CJ Queens Univ Dept Elect & Comp Engn Kingston ON K7L 3N6 Canada

Konstantinides and Yao have considered the problem of rank determination by use of effective singular values. In this correspondence, we show how to use the minimum description length criterion of Rissanen to provide an alternative means of estimating the index of the smallest nonzero singular value of a matrix when given estimates of the singular values.

关键词： Finite impulse response filter Acoustic noise Speech enhancement linear predictive coding Signal to noise ratio Speech analysis Speech synthesis Band pass filters Signal processing Speech coding

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Joint MELP turbo code with unequal error protection

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ELECTRONICS LETTERS 2001年第10期37卷 637-639页

作者： Wang, Q Koh, SN Nanyang Technol Univ Sch Elect & Elect Engn Singapore 639798 Singapore

The mixed excitation linear prediction (MELP) algorithm has been recently selected as the new federal standard for 2.4kbit/s coding of speech signals. The authors exploit the average residual inter-frame correlation and the error sensitivities of the bits in a MELP frame to enhance the robustness of the proposed joint MELP turbo coding schemes for Operations over Rayleigh fading channels.

关键词： BER speech coding mixed excitation linear prediction algorithm error statistics error detection combined source-channel coding turbo codes Codes unequal error protection correlation theory linear predictive coding residual inter-frame correlation Rayleigh channels bit error sensitivities Speech and audio coding Rayleigh fading channels Information theory joint MELP turbo code 2.4 kbit/s error correction MELP algorithm speech signals

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QUALITY IMPROVEMENT OF LPC-PROCESSED NOISY SPEECH BY USING SPECTRAL SUBTRACTION

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1989年第6期37卷 939-942页

作者： KANG, GS FRANSEN, LJ Naval Research Laboratory Inc. Washington D.C. DC USA

Numerous noise-suppression techniques have been developed for operating at the front end of low-bit-rate digital voice terminals. Some of these techniques have been evaluated by standardized intelligibility tests such as the diagnostic rhyme test (DRT). It is well known that the use of a noise suppressor seldom improves the DRT score even though listeners have had the impression that speech quality was enhanced. Unfortunately, noise suppressors have only occasionally been evaluated by standardized quality tests. The authors supplement quality test data for reference purposes. They use the diagnostic acceptability measure (DAM) to evaluate speech quality of the latest 2400-b/s linear-predictive coder (LPC) with a noise suppressor at the front end. They used a spectral subtraction technique for noise suppression. Ten different sets of noisy speech recorded at actual military platforms (such as a helicopter, tank, turboprop, helicopter carrier, or jeep) were input sources. The magnitude of the DAM improvement is substantial: as much as six points on the average, which is large enough to upgrade speech quality somewhat.< >

关键词： Signal processing algorithms Testing Speech enhancement Digital signal processing Noise measurement Speech analysis linear predictive coding Subtraction techniques Helicopters Acoustics

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A SPEECH SYNTHESIS DEVICE USING FORMANT AND RESIDUAL INFORMATION

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1983年第3期29卷 217-225页

作者： MITOME, Y FUSHIKIDA, K C&C Systems Research Laboratories NEC Corporation Limited Kawasaki Kanagawa Japan

A new speech synthesis device has been developed. This speech synthesizer is based on a formant synthesizer and uses residual information. Generally, synthesized speech using residual information has high quality. In this speech synthesizer, the residual information is introduced as FIR (Finite Impulse Response) filter coefficients. Great flexibility in bit rate can be obtained by this implementation. Speech synthesis experiment showed the flexibility and the high performance for this speech synthesizer. This speech synthesizer can be applied to a high quality text to speech synthesis system and many other systems.

关键词： Speech synthesis Finite impulse response filter Speech analysis Bit rate Information filtering Information filters Frequency synthesizers Speech processing Pulse modulation linear predictive coding

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