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IEEE Digital Signal Processing Workshop

作者： Tatsuhiko Kinjo Keiichi Funaki Graduate School of Science and Engineering University of Ryukyus Okinawa Japan Computing & Networking Center University of Ryukyus Okinawa Japan

ISBN: (纸本)1424405343

This paper proposes a novel robust fundamental frequency (F0) estimation algorithm based on complex-valued speech analysis for an analytic speech signal. Since analytic signal provides spectra only over positive frequencies, spectra can be accurately estimated in low frequencies. Consequently, it is considered that F0 estimation using the residual signal extracted by complex-valued speech analysis can perform better for F0 estimation than that for the residual signal extracted by conventional real-valued LPC analysis. In this paper, the autocorrelation function weighted by AMDF is adopted for the F0 estimation criterion and four signals; speech signal, analytic speech signal, LPC residual and complex LPC residual, are evaluated for the F0 estimation. Speech signals used in the experiments were corrupted by adding white Gaussian noise whose noise levels are 10, 5, 0, -5 [dB]. The experimental results demonstrate that the proposed algorithm based on complex speech analysis can perform better than other methods in an extremely noisy environment

关键词： Speech analysis Frequency estimation Signal analysis linear predictive coding Low-frequency noise Robustness Algorithm design and analysis Performance analysis Autocorrelation Speech enhancement

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On the Investigation of Spectral Resolution Problem for Identification of Female Speakers in Bengali

On the Investigation of Spectral Resolution Problem for Iden...

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IEEE International Conference on Industrial Technology (ICIT)

作者： Hemant A. Patil P. K. Dutta T. K. Basu Department of Electronics and Instrumentation Engineering Dr. B. C. Roy Engineering College Durgapur West Bengal India Department of Electrical Engineering Indian Institute of Technology Kharagpur West Bengal India

Voice biometrics is an economic method of person authentication with the help of machines because of low cost and high power computers. In this paper, we investigate the problem of spectral resolution in female speech for speaker identification. Finally, a speaker recognition system is presented to compare the relative performance of different LP-based features such as (LPC and LPCC) and filterbank-based features such as Mel-frequency cepstral coefficients (MFCC) for identification of female speakers. The results are shown for database collected from 15 female speakers in Bengali.

关键词： Biometrics Power system economics Power generation economics Authentication Costs Speech Speaker recognition linear predictive coding Cepstral analysis Mel frequency cepstral coefficient

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linear predictive speech coding using fermat number transform

Linear predictive speech coding using fermat number transfor...

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IEEE Region 8 International Symposium on Video/Image Processing and Multimedia Communications (EURASIP)

作者： G. Madre E.H. Baghious S. Azou G. Burel Laboratoire dElectronique et Systêmes de Télécommunications UMR CNRS 6165 Brest France Laboratoire d'Elcctronique et Systèmes de Télécommunications FRANCE

This paper is about the reduction of the computational complexity of a speech codec. A linear predictive coding procedure is developed to allow its implementation with number theoretic transforms. The use of fermat number transform can reduce, in a significant way, the cost of linear predictive algorithm implantation on digital signal processor.

关键词： Speech coding Speech analysis linear predictive coding Signal processing algorithms Speech synthesis Signal analysis Nonlinear filters Computational complexity Signal processing Performance analysis

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Voice quality evaluation in wireless packet communication systems: A tutorial and performance results for ROHC

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IEEE WIRELESS COMMUNICATIONS 2005年第1期12卷 60-67页

作者： Rein, S Fitzek, FHP Reisslein, M Aalborg Univ Dept Commun Technol Aalborg Denmark Arizona State Univ Dept Elect Engn Tempe AZ 85287 USA

As wireless systems evolve toward supporting a wide array of services, including traditional voice service, using packet-switched transport, it becomes increasingly important to assess the impact of packet-switched transport protocols on voice quality. In this article we present a tutorial on voice quality evaluation for wireless packet-switched systems. We introduce an evaluation methodology that combines elementary objective voice quality metrics with a frame synchronization mechanism. ne methodology allows networking researchers to conduct effective and accurate quality evaluation of packet voice. To illustrate the use of the described evaluation methodology and interpretation of the results, we conduct a case study of the impact of robust header compression (ROHC) on the voice quality achieved with real-time transmission of GSM encoded voice over a wireless link.

关键词： Tutorial Wireless communication GSM Decoding linear predictive coding Speech analysis Wireless application protocol Performance gain Internet Multimedia systems

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The 1974 origins of VoIP

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IEEE SIGNAL PROCESSING MAGAZINE 2005年第4期22卷 87-90页

作者： Gray, RM Massachusetts Institute of Technology University of Southern California

This paper brings light on the digital signal processing (DSP) roots of a modern concept, voice over IP (VoIP). An example is also provided in which developments in DSP - speech coding, in particular - had a profound impact on the early development of the ARPANET, the ancestor of the Internet. The author shows how packet speech, recently rediscovered and made popular as VoIP, was first successfully demonstrated in 1974 on the ARPANET and how the Internet protocol (IP) emerged largely as a result of that effort.

关键词： Internet telephony Digital signal processing History Speech coding linear predictive coding Speech processing Image coding Collaboration ARPANET predictive models

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Transforming real linear prediction coefficients to line spectral representations with a real FFT

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2005年第5期13卷 733-740页

作者： Yedlapalli, SS Texas Instruments India Pvt Ltd Bangalore 560017 Karnataka India

This paper describes a novel algorithm for transforming linear prediction coefficients (LPCs) to line spectral frequencies (LSFs) and line spectral pairs (LSPs) used by most of the speech processing applications. The symmetric and antisymmetric polynomials (SAPS) for LSP/LSFs, corresponding to the LPC polynomial, are first multiplexed into a single real sequence. The required samples of SAPS correspond to the DFT of the obtained real sequence. The proposed algorithm is referred as PMLS as it is based on the Plus Minus (PM) algorithm an FFT which efficiently computes the DFT of a real sequence for the positive frequency interval only. The samples of the SAPS are efficiently utilized for the computation of a single parameter which is used for computation of LSF and LSP independently with some interpolation principles. This interpolation exploits the available samples of SAPS and does not require their samples at finer resolution. The efficiency of the PMLS is illustrated with the help of some examples. Some guidelines for an optimal implementation of PMLS on fixed point digital signal processors (DSPs) are also presented.

关键词： discrete Fourier transforms discrete transforms interpolation iterative methods level-crossing problems linear predictive coding polynomial approximation spectral analysis

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Low-complexity lossless compression of hyperspectral imagery via linear prediction

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IEEE SIGNAL PROCESSING LETTERS 2005年第2期12卷 138-141页

作者： Rizzo, F Carpentieri, B Motta, G Storer, JA Univ Salerno Dipartimento Informat & Applicaz Renato M Capocel I-84081 Baronissi SA Italy Brandeis Univ Dept Comp Sci Waltham MA 02454 USA

We present a new low-complexity algorithm for hyperspectral image compression that uses linear prediction in the spectral domain. We introduce a simple heuristic to estimate the performance of the linear predictor from a pixel spatial context and a context modeling mechanism with one-band look-ahead capability, which improves the overall compression with marginal usage of additional memory. The proposed method is suitable to spacecraft on-board implementation, where limited hardware and low power consumption are key requirements. Finally, we present a least-squares optimized linear prediction technique that achieves better compression on data cubes acquired by the NASA JPL Airborne Visible/Infrared Imaging Spectrometer (AVIRIS).

关键词： data compression linear predictive coding remote sensing 3-D data

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Sultan of sound

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IEEE SPECTRUM 2005年第5期42卷 44-48页

作者： Perry, TS National Academy of Sciences National Academy of Engineering American Association for the Advancement of Science United States

This paper describes the pioneering research in the field of speech technology by James L. Flanagan, 2005 IEEE Medal of Honor awardee. Flanagan's work with speech coding heralded a series of advances over the years, including a currently favored technique, linear predictive coding. After graduating from Mississippi State as an electrical engineering major, Flanagan accepted a graduate assistantship in MIT's acoustics lab, which led to his seminal research in voice coding. Flanagan then worked at Bell Telephone Laboratories where he would spend the next 33 years. He climbed steadily up the ranks at Bell Labs, eventually becoming director of the Information Principles Research Laboratory. Among the projects that Flanagan was deeply involved in were the development of automatic speech recognition systems, voice mail, artificial larynx, and packet-switched voice technology.

关键词： Laboratories Medals Speech coding linear predictive coding Electrical engineering Acoustics Telephony Automatic speech recognition Voice mail Larynx

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A low-complexity doubletalk detector for acoustic echo cancellers in packet-based telephony

A low-complexity doubletalk detector for acoustic echo cance...

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Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Gordy, JD Goubran, RA Carleton Univ Dept Syst & Comp Engn Ottawa ON K1S 5B6 Canada

ISBN: (纸本)0780391543

This paper investigates a normalized cross-correlation-based doubletalk detector for acoustic echo cancellers. A novel low-complexity version is presented suitable for implementation on IP-enabled telephones employing LPC-based speech coders. In particular. the algorithm obtains a decorrelated input signal and decorrelation filter coefficients directly from the speech decoder. The proposed algorithm is implemented using ITU G.729. and simulation results are collected for a typical room. Calibration data are obtained and shown to be independent of the speech input signal. It is also shown that the proposed algorithm has the same performance as a full-complexity version employing the same decorrelation filter order.

关键词： Internet telephony decorrelation echo suppression filtering theory linear predictive coding speech coding vocoders IP-enabled telephones ITU G.729 LPC acoustic echo cancellers decorrelation filter coefficients low-complexity doubletalk detector packet-ba

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LPC Prediction Error Combined with LSB Steganography for Blind Speech Authentication

LPC Prediction Error Combined with LSB Steganography for Bli...

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IEEE Workshop on Multimedia Signal Processing

作者： Chai Peiqi Liu Jingyong Pei Dingyu Yang Zhiqiang Department of Computer Science University of Tongji Shanghai China

In this paper, we propose a new blind speech authentication method to verify the authenticity and integrity of speech. In this method, linear predictive coding (LPC) and least significant bit (LSB) steganography are combined, using LPC prediction error (LPCPE) as invariable features of speech and inserting these features into least significant bits of speech samples. When verifying the speech, we extract the inserted features, and compare them with recalculated LPCPE. Experimental results show that this method can detect not only the tampering in speech but also the position and region of the tampering

关键词： linear predictive coding Steganography Authentication Watermarking Speech coding Speech analysis Speech enhancement Computer errors Protection Speech synthesis

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