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

作者： Minsheng Liu A. Lacroix Institut fur Angewandte Physik University of Frankfurt am Main Frankfurt Germany

This paper presents the extended tube model for the vocal and nasal tract, in which the velum is described by means of a three-port-adaptor, which can be reduced to a two-port-adaptor. In this way the transfer function is derived, then its properties are investigated, so that this model can be used for the speech analysis of the nasal and non-nasal sounds and can be simply implemented by the inverse filtering method. Moreover the losses in the vocal and nasal tract are analysed and the shape of the nasal tract is also estimated.

关键词： Speech analysis Transfer functions Pharynx Acoustic scattering Filtering linear predictive coding Resonance Filters Shape Signal analysis

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A simple and efficient two-band speech coder at 2.4 kbit/s for real-time implementation on a single low-cost DSP

A simple and efficient two-band speech coder at 2.4 kbit/s f...

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Mediterranean Electrotechnical Conference (MELECON)

作者： F. Beritelli S. Casale F. Spataro Istituto di Informatica e Telecomunicazioni Universita Catania Catania Italy

In the new generation of very low bit rate speech coding models, multiband excitation (MBE) represents one of the most promising techniques. The paper proposes a new low-complexity two-band coding scheme capable of producing good speech quality at 2.4 kbit/s. Two new solutions are proposed to guarantee pitch continuity and establish the frequency of transition between the voiced and unvoiced portions of band. During informal listening tests we have seen that the speech coder proposed maintains a natural quality, perfect intelligibility and speaker identity. Finally, to demonstrate the simplicity of the algorithms proposed, the paper presents an evaluation of the speech coder complexity for a real-time implementation on a single low-cost digital signal processor.

关键词： Bit rate Speech coding Frequency Vocoders Speech analysis Speech processing Digital signal processors Speech synthesis linear predictive coding Fourier transforms

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A hybrid input/output spectrum adaptation scheme for LD-CELP coding of speech

A hybrid input/output spectrum adaptation scheme for LD-CELP...

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

作者： Siu-Pun Chui Cheung-Fat Chan Department of Electronic Engineering City University of Hong Kong Hong Kong China

ISBN: (纸本)0780331923

This paper describes a hybrid input/output block adaptation scheme for the quantization of spectral parameters for low-delay CELP coders. In this scheme, line spectrum pair (LSP) coefficients are used as parameters for spectral adaptation and they are partitioned into two segments; LSPs in the first segment which corresponding to low frequency regions are backward adapted from the reproduction speech while LSPs in the second segment are adapted from input speech and quantized as side information to the decoder. The rationale behind this scheme is to compensate for the spectrum distortion introduced by conventional backward adaptation schemes operating at low bit rates. Simulation results show that the hybrid LSP adaptation scheme achieves transparent quantization at 12 bits for each analysis frame of 5 ms.

关键词： Speech coding Delay Quantization Bit rate Speech analysis Acoustic noise Speech processing Speech enhancement linear predictive coding Frequency

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A low delay sub-band audio coder (20 Hz-15 kHz) at 64 kbit/s

A low delay sub-band audio coder (20 Hz-15 kHz) at 64 kbit/s

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IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： K. Hay L. Mainard S. Saoudi ENST-Br Department SC. Technopôle Brest-Iroise Brest France CCETT Servive RCS/SDA France

a new method for coding generic audio signals at 64 kbit/s in the bandwidth 20-15000 Hz with a low delay is presented. It combines sub-band coding, low delay CELP algorithm and cascaded filterbanks. We show how the different parameters of LD-CELP can be adapted to achieve a better quality, perceptual coding techniques are integrated into the encoder for allocating bits to each sub-band.

关键词： Delay Bandwidth Bit rate Decoding Psychology Psychoacoustic models linear predictive coding Speech coding Filters Sampling methods

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Objective speech quality measure for cellular phone

Objective speech quality measure for cellular phone

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

作者： K.H. Lam O.C. Au C.C. Chan K.F. Hui S.F. Lau Department of Electrical and Electronic Engineering Hong Kong University of Science and Technology Kowloon Hong Kong China

Cellular phone network speech quality monitoring is a regular task performed by the cellular service providers. Objective speech quality measures are needed in such tasks to provide a reasonably accurate estimate of subjective quality of the network. We performed an experiment to collect real distorted data, conducted a survey to obtain subjective quality measure of the collected speech samples and studied the statistical correlation of 32 objective speech quality measures with the subjective measures. Four of the objective measures were found to be good. Synchronization was found to be important.

关键词： Speech Cellular phones Distortion measurement Cellular networks Time measurement Auditory system Mean square error methods linear predictive coding Frequency measurement Psychoacoustic models

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Comparison of text-independent speaker recognition methods on telephone speech with acoustic mismatch

Comparison of text-independent speaker recognition methods o...

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International Conference on Spoken Language, ICSLP

作者： S. van Vuuren Center for Spoken Language Understanding Oregon Graduate Institute of Science and Technology Beaverton OR USA

We compare speaker recognition performance of vector quantization (VQ), Gaussian mixture modeling (GMM) and the Arithmetic Harmonic Sphericity measure (AHS) in adverse telephone speech conditions. The aim is to address the question: how do multimodal VQ and GMM typically compare to the simpler unimodal AHS for matched and mismatched training and testing environments? We study identification (closed set) and verification errors on a new multi environment database. We consider LPC and PLP features as well as their RASTA derivatives. We conclude that RASTA processing can remove redundancies from the features. We affirm that even when we use channel and noise compensation schemes, speaker recognition errors remain high when there is acoustic mismatch.

关键词： Speaker recognition Automated highways Vector quantization Arithmetic Telephony Speech Testing Spatial databases linear predictive coding Redundancy

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A 2.4 kbit/s MELP coder candidate for the new U.S. Federal Standard

A 2.4 kbit/s MELP coder candidate for the new U.S. Federal S...

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

作者： A. McCree Kwan Truong E.B. George T.P. Barnwell V. Viswanathan Texas Instruments Personal Systems Laboratory Dallas TX USA Atlanta Signal Processors Inc. Atlanta GA USA

This paper describes our enhanced mixed excitation linear prediction (MELP) speech coder which is a candidate for the new U.S. Federal Standard at 2.4 kbits/s. The new coder is based on the MELP model, and it uses a number of enhancements as well as efficient quantization algorithms to improve performance while maintaining a low bit rate. In addition, the coder has been optimized for performance in acoustic background noise and in channel errors, as well as for efficient real-time implementation. Listening tests confirm that the enhanced 2.4 kbit/s MELP coder performs as well as the higher bit rate 4.8 kbit/s FS1016 CELP standard.

关键词： Code standards linear predictive coding Fourier series Filters Quantization Bit rate Performance evaluation Speech synthesis Signal processing algorithms Vocoders

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Wavelet based feature extraction for phoneme recognition

Wavelet based feature extraction for phoneme recognition

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International Conference on Spoken Language, ICSLP

作者： C.J. Long S. Datta Department of Electronic and Electrical Engineering Loughborough University of Technology Loughborough UK

In an effort to provide a more efficient representation of the acoustical speech signal in the pre classification stage of a speech recognition system, we consider the application of the Best-Basis Algorithm of R.R. Coifman and M.L. Wickerhauser (1992). This combines the advantages of using a smooth, compactly supported wavelet basis with an adaptive time scale analysis, dependent on the problem at hand. We start by briefly reviewing areas within speech recognition where the wavelet transform has been applied with some success. Examples include pitch detection, formant tracking, phoneme classification. Finally, our wavelet based feature extraction system is described and its performance on a simple phonetic classification problem given.

关键词： Feature extraction Multiresolution analysis Continuous wavelet transforms Discrete wavelet transforms Speech recognition Wavelet analysis Wavelet transforms Time frequency analysis Signal analysis linear predictive coding

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A MIXED EXCITATION LPC VOCODER MODEL FOR LOW BIT-RATE SPEECH coding

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1995年第4期3卷 242-250页

作者： MCCREE, AV BARNWELL, TP GEORGIA INST TECHNOL SCH ELECT ENGNATLANTAGA 30332

Traditional pitch-excited linear predictive coding (LPC) vocoders use a fully parametric model to efficiently encode the important information in human speech. These vocoders can produce intelligible speech at low data rates (800-2400 b/s), but they often sound synthetic and generate annoying artifacts such as buzzes, thumps, and tonal noises. These problems increase dramatically if acoustic background noise is present at the speech input. This paper presents a new mixed excitation LPC vocoder model that preserves the low bit rate of a fully parametric model but adds more free parameters to the excitation signal so that the synthesizer can mimic more characteristics of natural human speech. The new model also eliminates the traditional requirement for a binary voicing decision so that the vocoder performs well even in the presence of acoustic background noise. A 2400-b/s LPC vocoder based on this model has been developed and implemented in simulations and in a real-time system. Formal subjective testing of this coder confirms that it produces natural sounding speech even in a difficult noise environment. In fact, diagnostic acceptibility measure (DAM) test scores show that the performance of the 2400-b/s mixed excitation LPC vocoder is close to that of the government standard 4800-b/s CELP coder.

关键词： linear predictive coding Vocoders Bit rate Speech coding Parametric statistics Humans Acoustic noise Background noise Speech enhancement Acoustic testing

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DESIGN OF A PITCH SYNCHRONOUS INNOVATION CELP CODER FOR MOBILE COMMUNICATIONS

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IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS 1995年第1期13卷 31-41页

作者： MANO, K MORIYA, T MIKI, S OHMURO, H IKEDA, K IKEDO, J NIPPON TELEGRAPH & TEL PUBL CORP YOKOSUKA ELECT COMMUN LABS RADIO COMMUN SYST LABS YOKOSUKA KANAGAWA 23803 JAPAN

This paper describes the design of a speech coder called pitch synchronous innovation CELP (PSI-CELP) for low bit-rate mobile communications. PSI-CELP is based on CELP, but has more adaptive excitation structures. In voiced frames, instead of conventional random excitation vectors, PSI-CELP converts even the random excitation vectors to have pitch periodicity by repeating stored random vectors as well as by using an adaptive codebook. In silent, unvoiced, and transient frames, the coder stops using the adaptive codebook and switches to fixed random codebooks. The PSI-CELP coder also implements novel structures and techniques: an FIR-type perceptual weighting filter using unquantized LPC parameters, a random codebook with a conjugate structure trained to be robust against channel errors, codebook search with delayed decision, a gain quantization with sloped amplitude, and a moving average prediction coding of LSP parameters. Our speech coder is implemented by DSP chips. Its coded speech quality at 3.6 kb/s with 2.0 kb/s redundancy is comparable to that of the Japanese full-rate VSELP coder at 6.7 kb/s with 4.5 kb/s redundancy. The basic structure of this PSI-CELP coder has been chosen as the Japanese half-rate speech codec for digital cellular telecommunications.

关键词： Technological innovation Speech Redundancy Mobile communication Switches Filters linear predictive coding Robustness Delay Quantization

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