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VQ CODEBOOK DESIGN ALGORITHM-BASED ON PARTIAL GLA

ELECTRONICS LETTERS 1995年第21期31卷 1803-1805页

作者： CHEN, CQ KOH, SN SIVAPRAKASAPILLAI, P School of Electrical and Electronic Engineering Nanyang Technological University Singapore Republic of Singapore

A novel scheme of generating the codebook for vector quantisation is presented. With the initial codebook resulting from a K-d tree splitting procedure based on the greatest coordinate variance, a proposed partial GLA is used to improve the codevectors. The performance of the VQ so obtained is superior to those of the VQ designed by the standard GLA with the same initialisation and the splitting-initialised LEG algorithm. However, the improvement in performance is accompanied by an increase in the computational complexity involved in the designed stage.

关键词： VECTOR QUANTIZATION linear predictive coding

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EFFICIENT SCALAR QUANTIZATION OF LINE SPECTRUM PAIRS USING THE ORDERING PROPERTY

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ELECTRONICS LETTERS 1995年第8期31卷 611-612页

作者： LEE, I WOO, HC KANG, S TAEGU UNIV DEPT ELECTR ENGNTAEGUSOUTH KOREA HANYANG UNIV DEPT CONTROL ENGNSEOULSOUTH KOREA

An efficient quantisation method of line spectrum pairs (LSP) which has good performance and very low complexity and memory is proposed. The ordering property of the LSP parameters is utilised in the DPCM scheme. The new scalar quantisation algorithm requires 32 bit/frame to achieve 1 dB(2) average spectral distortion. The quantisation performance has also been shown to be robust across databases and different speakers.

关键词： linear predictive coding SPEECH coding

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linear PREDICTION USING L(1) NORM IN ORTHOGONAL VECTOR-SPACE

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ELECTRONICS LETTERS 1995年第6期31卷 430-431页

作者： HU, HT Department of Electronic Engineering National I-Lan Institute of Agriculture and Technology I-Lan Republic of China

linear prediction is formulated in a vector space by means of the orthogonal transformation, with which the L(1) criterion can be easily incorporated to yield an efficient iterative algorithm. An improvement over the ... 详细信息

关键词： linear predictive coding SPEECH ANALYSIS AND PROCESSING

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SPEECH CLASSIFICATION EMBEDDED IN ADAPTIVE CODEBOOK SEARCH FOR LOW BIT-RATE CELP coding

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1995年第1期3卷 94-98页

作者： KUO, CC JEAN, FR WANG, HC Department of Electrical Engineering National Tsing Hua University Hsinchu Taiwan

This correspondence proposes a new CELP coding method which embeds speech classification in adaptive codebook search. This approach can retain the synthesized speech quality at bit-rates below 4 kb/s. A pitch analyzer is designed to classify each frame by its periodicity, and with a finite-state machine, one of four states is determined. Then the adaptive codebook search scheme is switched according to the state. Simulation results show that higher SEGSNR and lower computation complexity can be achieved, and the pitch contour of the synthesized speech is smoother than that produced by conventional CELP coders.

关键词： Speech coding Speech analysis Speech synthesis Decoding Computational modeling Delay linear predictive coding Speech recognition Councils Filters

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THE IMPACT OF VOICE PROCESSING ON MODERN TELECOMMUNICATIONS

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SPEECH COMMUNICATION 1995年第3-4期17卷 217-226页

作者： RABINER, LR AT&T BELL LABS INFORMAT PRINCIPLES RES LABS 600 MT AVE MURRAY HILL NJ 07974 USA

Research has been conducted in the area of voice processing for over six decades but it has only been in the past few years that the impact of the years of research is starting to be seen in modern telecommunications systems. Virtually every area of voice processing, including speech coding, speech synthesis, speech recognition, and even, to a small extent, speaker verification, has left the research laboratory and now appears in a product or service that is in daily use out in the marketplace, often by millions of customers per day. This revolution in voice processing in telecommunications is fueled by algorithmic advances (which improve the quality of the voice processing systems), hardware advances (which provide high processing power and memory at low cost), and networking advances (which provide high bandwidth pipes to the home, office, and throughout the telecommunications network). In this paper we illustrate the impact of voice processing on modern telecommunications by showing the diverse ways in which speech coding, speech synthesis, speech recognition and speaker verification have become embodied in new products and services.

关键词： Speech coding linear predictive coding Code excited linear prediction Voice messaging systems Voice response systems Telephone security devices Telephone bandwidth coders Wideband speech coders Digital audio broadcast Speech synthesis Text-to-speech synthesis Speech recognition Template systems Hidden Markov models Isolated word recognition Connected word recognition Continuous speech recognition Speaker verification

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A COMPARISON OF SIGNAL-PROCESSING FRONT-ENDS FOR AUTOMATIC WORD RECOGNITION

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1995年第4期3卷 286-293页

作者： JANKOWSKI, CR VO, HDH LIPPMANN, RP Department of Electrical Engineering and Computer Science Massachusetts Institute of Technology Lexington MA USA Lincoln Laboratory Massachusetts Institute of Technology Lexington MA USA

This paper compares the word error rate of a speech recognizer using several signal processing front ends based on auditory properties. Front ends were compared with a control mel filter bank (MFB) based cepstral front end in clean speech and with speech degraded by noise and spectral variability, using the TI-105 isolated word database. MFB recognition error rates ranged from 0.5 to 26.9% in noise, depending on the SNR, and auditory models provided error rates as much as four percentage points lower. With speech degraded by linear filtering, MFB error rates ranged from 0.5 to 3.1%, and the reduction in error rates provided by auditory models was less than 0.5 percentage points. Some earlier studies that demonstrated considerably more improvement with auditory models used linear predictive coding (LPC) based control front ends. This paper shows that MFB cepstra significantly outperform LPC cepstra under noisy conditions. Techniques using an optimal linear combination of features for data reduction were also evaluated.

关键词： Signal processing Error analysis linear predictive coding Speech enhancement Degradation Speech processing Speech recognition Automatic speech recognition Automatic control Filter bank

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A speech coding algorithm based on predictive coding

A speech coding algorithm based on predictive coding

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Data Compression Conference (DCC)

作者： S. Kwong K.F. Man Department of Computer Science City University of Hong Kong Hong Kong China

Summary form only given. A compression algorithm for high quality speech signal using predictive coding techniques is developed. Code-excited linear predictive coding (CELPC) is one of the key techniques to compress speech signal to a bit-rate around 4.8 Kbps. However, due to the heavy computational requirement in the CELPC and speech signals usually can be divided into two portions: namely the based-band and the high-band frequency range. A hybrid CELPC and voice excited linear predictive coding (VELPC) scheme is developed for speech coding to reduce the complexity of the original CELPC. In the algorithm, a speech signal is firstly divided into two portions, the based-band and high-band respectively, in frequency domain, and then the low portion is coded with CELPC and the high-band portion is coded with VELPC. The test experiments showed this new coder can produce synthesized speech with good quality at a better bit rates than the original CELPC. When using the coding methods for the base-band and the high-band signal, we must decide how to divide the speech signal into two portions. In choosing the bandwidth of the base-band signal, there is a trade-off between the coding quality and the bit rate. In our experiment, the bandwidth of the base-band signal is chosen as one fourth of that of the original speech. Subjective evaluation experiments were conducted to test the performance of the hybrid CELPC and VELPC technique. For speech signal sampled at 8 kHz, a bit rate of 4.0 kbps can be achieved with frame intervals of 23 ms. The experimental results showed that the quality of the synthesized speech using hybrid coding technique at the bit rate of 4.0 kbps was almost the same as that of the CELPC at the bit rate of 4.8 kbps.

关键词： Speech coding Prediction algorithms predictive coding Bit rate Speech synthesis linear predictive coding Testing Signal synthesis Bandwidth Compression algorithms

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A Two Codebook Format for Robust Quantization of Line Spectral Frequencies

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IEEE Transactions on Speech and Audio Processing 1995年第3期3卷 157-168页

作者： Ramachandran, Ravi P. Sondhi, Man Mohan Seshadri, Nambi Atal, Bishnu S. CAIP Center Department of Electrical Engineering Rutgers University Piscataway NJ United States AT&T Bell Laboratories Murray Hill NJ 07974 United States

An important problem in speech coding is the quantization of linear predictive coefficients (LPC) with the smallest possible number of bits while maintaining robustness to a large variety of speech material and transmission media. Since direct quantization of LPC’s is known to be unsatisfactory, we consider this problem for an equivalent representation, namely, the line spectral frequencies (LSF). To achieve an acceptable level of distortion a scalar quantizer for LSF’s requires a 36 bit codebook. We derive a 30 bit two-quantizer scheme which achieves a performance equivalent to this scalar quantizer. This equivalence is verified by tests on data taken from various types of filtered speech, speech corrupted by noise and by a set of randomly generated LSF’s. The two-quantizer format consists of both a vector and a scalar quantizer such that for each input, the better quantizer is used. The vector quantizer is designed from a training set that reflects the joint density (for coding efficiency) and which ensures coverage (for robustness). The scalar quantizer plays a pivotal role in dealing better with regions of the space that are sparsely covered by its vector quantizer counterpart. A further reduction of 1 bit is obtained by formulating a new adaptation algorithm for the vector quantizer and doing a dynamic programming search for both quantizers. The method of adaptation takes advantage of the ordering of the LSF’s and imposes no overhead in memory requirements. The dynamic programming search is feasible due to the ordering property. Subjective tests in a speech coder reveal that the 29 bit scheme produces equivalent perceptual quality to that when the parameters are unquantized. © 1995 IEEE

关键词： Robustness Quantization Frequency linear predictive coding Speech analysis Filters Speech coding Testing Speech enhancement Dynamic programming

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LOW-DELAY HYBRID VECTOR EXCITATION linear predictive SPEECH coding

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ELECTRONICS LETTERS 1993年第25期29卷 2164-2165页

作者： CHEN, H WONG, WC KO, CC Department of Electrical Engineering National University of Singapore Singapore

A hybrid approach in determining the excitation vector in a low-delay code excited linear predictive coder is proposed. By a judicious division of the composite excitation vector into long-term and short-term componen... 详细信息

关键词： linear predictive coding SPEECH coding

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Speaker identification based on nonlinear speech models

Speaker identification based on nonlinear speech models

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Asilomar Conference on Signals, Systems & Computers

作者： S. Wenndt S. Shamsander Rome Laboratory/IRAA Rome NY USA

Some of the work on speech processing has focused on modeling speech as an AM-FM signal. The success of the AM-FM model motivated us to investigate a similar nonlinear model and examine its application in speaker identification. Tests are carried out to compare the performance of the novel cyclic correlation based method with popular speaker identification methods based on cepstra. These studies show that the performance of the proposed method is comparable to the cepstrum based approach at high signal-to-noise ratio, but the former outperforms the latter under noisy conditions.

关键词： Testing Databases Cepstrum Signal to noise ratio Telephony Noise reduction Speech processing linear predictive coding Cepstral analysis Noise robustness

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