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A fast LSF search algorithm based on interframe correlation in G.723.1

EURASIP JOURNAL ON APPLIED SIGNAL PROCESSING 2004年第8期2004卷 1107-1112页

作者： Kibey, SA Kulkarni, JP Sarode, PD Tata Elxsi Ltd Digital Signal Proc & Multimedia Grp Bangalore 560048 Karnataka India Indian Inst Sci Ctr Elect Design & Technol Bangalore 560012 Karnataka India Honeywell Technol Solut Labs Pvt Ltd Bangalore 560076 Karnataka India

We explain a time complexity reduction algorithm that improves the line spectral frequencies (LSF) search procedure on the unit circle for low bit rate speech codecs. The algorithm is based on strong interframe correlation exhibited by LSFs. The fixed point C code of ITU-T Recommendation G.723.1, which uses the "real root algorithm" was modified and the results were verified on ARM-7TDMI general purpose RISC processor. The algorithm works for all test vectors provided by International Telecommunications Union-Telecommunication (ITU-T) as well as real speech. The average time reduction in the search computation was found to be approximately 20%.

关键词： line spectral frequencies linear predictive coding unit circle interframe correlation G.7.23.1

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Underwater target classification using wavelet packets and neural networks

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IEEE TRANSACTIONS ON NEURAL NETWORKS 2000年第3期11卷 784-794页

作者： Azimi-Sadjadi, MR Yao, D Huang, Q Dobeck, GJ Colorado State Univ Signal Image Proc Lab Dept Elect & Comp Engn Ft Collins CO 80523 USA NSWC Dahlgren Div Coastal Syst Stn Panama City FL 32407 USA

In this paper, a new subband-based classification scheme is developed for classifying underwater mines and mine-like targets From the acoustic backscattered signals. The system consists of a feature extractor using wavelet packets in conjunction with linear predictive coding (LPC), a feature selection scheme, and a backpropagation neural-network classifier, The data set used for this study consists of the backscattered signals from six different objects: two mine-like targets and four nontargets for several aspect angles. Simulation results on ten different noisy realizations and for signal-to-noise ratio (SNR) of 12 dB are presented. The receiver operating characteristic (ROC) curve of the classifier generated based on these results demonstrated excellent classification performance of the system, The generalization ability of the trained network was demonstrated by computing the error and classification rate statistics on a large data set, A multiaspect fusion scheme was also adopted in order to further improve the classification performance.

关键词： feature extraction linear predictive coding neural network underwater target classification wavelet packets

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Formant Location From LPC Analysis Data

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第2期1卷 129-134页

作者： Snell, Roy C. Milinazzo, Fausto Royal Roads Mil Coll Dept Math Victoria BC V0S 1B0 Canada

The estimation of formant frequencies and bandwidths from the filter coefficients obtained through LPC analysis of speech is discussed from several viewpoints. A new method for locating roots within the unit circle is derived. This algorithm is particularly well suited to computations carried out in fixed point arithmetic using specialized signal processing hardware.

关键词： linear predictive coding Data analysis Signal processing algorithms Speech analysis Digital signal processing chips Frequency Fixed-point arithmetic Resonance Polynomials Filters

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A TIME-VARYING ANALYSIS METHOD FOR RAPID TRANSITIONS IN SPEECH

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第4期39卷 815-824页

作者： NATHAN, KS LEE, YT SILVERMAN, HF BROWN UNIV DIV ENGN ENGN MAN MACHINE SYST LAB PROVIDENCE RI 02912 USA BELLCORE MORRISTOWN NJ 07960 USA

Although many time-varying models have been proposed in the speech literature, few, if any, have had success when applied to speech in real applications. This is in part due to the limitations of the models themselves as well as the lack of robustness of their attendant parameter extraction methods. An LPC model based on time-dependent poles has yielded promising results when applied to synthetic data. This paper extends this technique to real speech data. The data are processed pitch synchronously using a simple procedure to identify regions of the data that best fit the model. The maximum-likelihood technique, which has been found to be robust in the presence of noise, is used to estimate the parameters. Resulting format estimates for several diphthongs are presented;our algorithm tracks the formants well, both in stable regions and in regions of transition. This ability to track formant variation within analysis intervals is a definite advantage over traditional LPC. Results from speech data involving final stop consonants are presented. Rapid changes, particularly in the first and second formants, in the region immediately prior to the stop have been detected. Such abrupt transistions are often not detected by traditional time-invariant methods. This information about the behavior of formants prior to the stop will be useful as additional information for characterization of stop consonants.

关键词： Speech analysis Parameter estimation linear predictive coding Maximum likelihood estimation Data mining Difference equations Parameter extraction Maximum likelihood detection Noise robustness Parametric statistics

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COMBINED SPEECH AND CHANNEL coding FOR MOBILE RADIO-COMMUNICATIONS

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1994年第4期43卷 1078-1087页

作者： SHI, H HO, PKM CUPERMAN, V Communications Science Laboratory School of Engineering Science Simon Fraser University Burnaby BC Canada

We present, in this paper, a combined speech and channel coding scheme for digital mobile communications. The speech coding algorithm is based on Code-Excited linear Prediction (CELP) and it provides good communication quality speech at a rate of 4 kbps. For the channel code, both rate-compatible punctured convolutional (RCPC) codes and punctured Reed-Solomon (PRS) codes are considered. In the case of RCPC codes, soft decision decoding is considered, in addition to the simpler hard decision decoding. The modulation format chosen in our study is pi/4-DQPSK with differential detection. Unequal error protection is used based on the bit error sensitivities of the different speech parameters. The performance of the combined speech and channel codec is studied under different mobile channel conditions, such as fade rates, signal-to-noise ratios, and interleaving delays. The results indicate that, with no interleaving delay and a large channel signal-to-noise ratio, PRS codes and RCPC codes with soft decision decoding have similar performance, given in terms of the segmental signal-to-noise ratio (SSNR) of the reconstructed speech. At low channel SNR, RCPC codes with soft decision decoding are significantly better than PRS codes. Informal listening tests were also conducted and the results indicate that the difference between the quality of the reconstructed speech in an error-free channel and that in a Rayleigh fading channel is imperceptible at channel SNR's larger than 20 dB for both the RCPC and the PRS-based codecs.

关键词： Rayleigh channels Reed-Solomon codes channel coding convolutional codes decoding demodulation differential phase shift keying digital radio fading land mobile radio linear predictive coding quadrature phase shift keying speech codecs speech coding speech

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Smelting condition identification for a fused magnesium furnace based on an acoustic signal

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JOURNAL OF MATERIALS PROCESSING TECHNOLOGY 2017年 244卷 231-239页

作者： Fu, You Wang, Ninghui Wang, Zhen Wang, Zhiqiang Ji, Bing Wang, Xiaochen Dalian Univ Technol Sch Elect Engn Dalian 116024 Peoples R China Dalian Univ Technol Sch Innovat & Entrepreneurship Dalian 116024 Peoples R China Univ Leicester Dept Engn Univ Rd Leicester LE1 7RH Leics England Environm Engn Assessment Ctr Liaoning Prov Shenyang 110032 Peoples R China

To promote energy efficiency during *** furnace smelting, four smelting states were introduced in the smelting stage: an unmelted state, semi-molten state, molten state, and overheating state. A smelting identification system to distinguish these smelting states was developed through the use of linear predictive coding and a principal component analysis algorithm. A new smelting condition identification system was obtained. Corresponding pilot productions were conducted to compare the differences between employing the method and not employing the method. All of the pilot production data showed that feeding raw materials over time during the overheating state and decreasing current injection in the molten state could reduce energy consumption as well as increase crystal purity. (C) 2016 Published by Elsevier B.V.

关键词： Acoustic signal Fused magnesium furnace linear predictive coding Principal component analysis

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VECTOR QUANTIZATION OF THE ARTICULATORY SPACE

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1988年第12期36卷 1812-1818页

作者： LARAR, JN SCHROETER, J SONDHI, MM AT&T BELL LABS ACOUST RES DEPTMURRAY HILLNJ 07974

A technique is presented for quantizing the articulatory space, i.e. for replacing the continuum of all possible vocal tract shapes by a finite set of shapes that span the articulatory space. Vocal tract shapes are represented as vectors of parameters that control an articulatory model, so vector quantization are applicable for deriving this codebook of tract shapes. Samples of the articulatory space are generated by interpolating model parameters along paths between selected target configurations. These shapes constitute the training data that are clustered to form the codebook. Such a codebook of shapes has direct application when vocal tract shapes must be estimated from speech. One such application, which motivated the present study, is the use of the codebook to providing starting configurations for an optimizing articulatory speech synthesizer.< >

关键词： Vector quantization linear predictive coding Speech analysis Synthesizers Shape control Speech synthesis Tongue Spectral shape Speech coding Books

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A VERY SIMPLE AND EFFICIENT WEIGHTING FILTER WITH APPLICATION TO A CELP CODER FOR HIGH-QUALITY SPEECH AT 4800 BITS/S

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SIGNAL PROCESSING 1992年第2期27卷 109-116页

作者： BOITE, R LEICH, H GAO, Y Laboratoire de Théorie des Circuits et Traitement des Signaux (T.C.T.S.) Faculté Polytechnique de Mons 33 Boulevard Dolez B-7000 Mons Belgium Lernout & Hauspie Speechproducts n.v. Rozendaalstraat 14 8900 Ieper Belgium

In this paper we propose a very simple and efficient weighting filter with which the computational complexity of CELP coders can be considerably reduced. Other algorithms using a weighting filter could also benefit from the advantages of this simplified weighting filter. The estimation of the long term prediction with the close-loop method is described. A binary codebook is used for the excitation vectors. It is shown how the excitation sequence can be obtained by a non-exhaustive method in two steps with a simplified algorithm and the simple weighting filter. Several coders have been implemented showing that the perceptual quality of the simplified algorithm is equivalent to that of the original CELP.

关键词： SIGNAL PROCESSING SPEECH coding linear predictive coding SPEECH CELP coding SPEECH PERCEPTION

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A TOLL QUALITY 8-KB/S SPEECH CODEC FOR THE PERSONAL COMMUNICATIONS-SYSTEM (PCS)

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1994年第3期43卷 808-816页

作者： SALAMI, R LAFLAMME, C ADOUL, JP MASSALOUX, D CTR NATL ETUD TELECOMMUN F-22301 LANNIONFRANCE

A toll quality speech codec at 8 Kb/s suitable for the future personal communications system is presented. The codec is currently under standardization by the ITU-T (successor of CCITT) where the codec terms of reference were mainly determined considering PCS application. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP) and has a speech frame of 10 ms. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Formal subjective listening tests, performed by ITU-T SG 12, showed that the codec quality is equivalent to that of G.726 ADPCM at 32 Kb/s in error-free conditions and it outperforms G.726 under error conditions. The codec performs adequately under tandeming conditions, and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements, and this is one subject of study for the next phase. The algorithm has been implemented on a single fixed-point DSP for the ITU-T subjective test, and required about 29 MIPS. A recent optimized version, however, requires 24 MIPS without any speech quality degradation,

关键词： analogue-digital conversion codecs linear predictive coding mobile radio systems personal communication networks speech coding telecommunication standards vocoders 8 kbit/s ACELP ITU-T subjective test PCS algebraic code-excited linear prediction codebook codecs telecommunication standards Personal communication networks ACELP speech codecs speech coding POWER CONVERSION SYSTEM Analog-digital conversion Vocoders Mobile radio systems Personal communication systems linear Prediction coding codebook

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Design and evaluation of a parametric model for cardiac sounds

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COMPUTERS IN BIOLOGY AND MEDICINE 2017年 89卷 170-180页

作者： Ibarra-Hernandez, Roilhi F. Alonso-Arevalo, Miguel A. Cruz-Gutierrez, Alejandro Licona-Chavez, Ana L. Villarreal-Reyes, Salvador Ctr Invest Cient & Educ Super Ensenada Div Fis Aplicada Dept Elect & Telecomunicac Carretera Ensenada Tijuana 3918 Ensenada 22860 Baja California Mexico Ctr Estudios Univ Xochicalco Fac Med Campus EnsenadaSan Francisco 1139 Ensenada 22830 Baja California Mexico

Heart sound analysis plays an important role in the auscultative diagnosis process to detect the presence of cardiovascular diseases. In this paper we propose a novel parametric heart sound model that accurately represents normal and pathological cardiac audio signals, also known as phonocardiograms (PCG). The proposed model considers that the PCG signal is formed by the sum of two parts: one of them is deterministic and the other one is stochastic. The first part contains most of the acoustic energy. This part is modeled by the Matching Pursuit (MP) algorithm, which performs an analysis-synthesis procedure to represent the PCG signal as a linear combination of elementary waveforms. The second part, also called residual, is obtained after subtracting the deterministic signal from the original heart sound recording and can be accurately represented as an autoregressive process using the linear predictive coding (LPC) technique. We evaluate the proposed heart sound model by performing subjective and objective tests using signals corresponding to different pathological cardiac sounds. The results of the objective evaluation show an average Percentage of Root-Mean-Square Difference of approximately 5% between the original heart sound and the reconstructed signal. For the subjective test we conducted a formal methodology for perceptual evaluation of audio quality with the assistance of medical experts. Statistical results of the subjective evaluation show that our model provides a highly accurate approximation of real heart sound signals. We are not aware of any previous heart sound model rigorously evaluated as our proposal.

关键词： Heart sounds modeling Biomedical signal processing Matching Pursuit Time-frequency dictionaries Time-frequency atoms linear predictive coding

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