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Asia-Pacific Conference on Circuits and Systems

作者： Dutono, T Watanabe, A ITS EEPIS Surabaya Indonesia

ISBN: (纸本)0780376900

Currently, speech recognition technology is in a mature condition, however, comparing with others applied technology, speech recognition technology still have limited daily live applications. One of the reasons is that the lack of our knowledge regarding the speech features. We performed word recognition experiments using features that is extracted by several different ways. We called them multi domain features. As shown in the results of recognition experiments, it is interesting that the multi domain parameters which represent phonemic features are very effective for the word recognition and operate to promote the goodness of the recognition in cooperation with the other conditions like a distance measure.

关键词： Cepstrum Dictionaries Euclidean distance Feature extraction Filter bank Jamming linear predictive coding Pattern recognition Speech recognition Vocabulary

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On Creating Reference Templates for Speaker Independent Recognition of Isolated Words

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IEEE Transactions on Acoustics, Speech, and Signal Processing 1978年第1期26卷 34-42页

作者： Rabiner, Lawrence R. Bell Laboratories Murray Hill NJ 07974 United States

The three aspects of a statistical approach to a pattern recognition problem are the selection of features, choice of a measure of similarity, and a method for creating the reference templates (patterns) used in the statistical tests. This paper discusses a philosophy for creating reference templates for a speaker independent, isolated word recognition system. Although there remain many unanswered questions both about how to select appropriate features for recognition, and how to measure similarity between sets of features, such issues are not discussed here. Instead we concentrate on methods for creating the reference templates. In particular, a method of combining word patterns from a number of speakers is proposed in which a clustering type of analysis is used to determine which patterns are merged to create a word template. The creation of multiple templates, based on this method, is discussed and is shown to be of substantial value for as few as eight speakers in the training set. To test the ideas proposed here, a 54 word vocabulary word recognition system was implemented. All input words were recorded off a standard telephone line. The features used were the LPC coefficients of an 8-pole analysis, and the simple Itakura distance measure was used to measure similarity between patterns. With word templates obtained as described above, recognition accuracies of 85 percent were obtained in a forced choice recognition test on the 54 word vocabulary using eight new speakers. The correct word was within the top five choices 98 percent of the time. Using a strategy in which all the training words were used to create the templates, the recognition accuracy fell to 77 percent, and the correct word was within the top five choices only 89 percent of the time. © 1978 IEEE

关键词： linear predictive coding Pattern recognition Speech Energy measurement Frequency measurement Time measurement Filtering theory Band pass filters Frequency domain analysis Cepstral analysis

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Combined speech and audio coding by discrimination

Combined speech and audio coding by discrimination

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7th IEEE Workshop on Speech coding

作者： Tancerel, L Ragot, S Ruoppila, VT Lefebvre, R Univ Sherbrooke Dept Elect Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)0780364163

We propose in this paper a general solution for combined speech and audio coding. Particularly, we describe a speech/music discrimination procedure for multi-mode wideband coding. The speech/music decision is updated only when a low-energy frame is detected, and kept unchanged otherwise. The signal is classified using second-order statistics of discriminant parameters. An experimental CELP/transform coder operating at 16 kbit/s is demonstrated. Results show improved performance when compared to single-mode encoding.

关键词： Audio coding Bit rate Delay estimation Forward contracts linear predictive coding Multiple signal classification Speech coding Statistics Switches Wideband

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Automated tongue-twister phrase-based screening for Cerebellar Ataxia using Vocal tract Biomarkers 41

Automated tongue-twister phrase-based screening for Cerebell...

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41st Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： Kashyap, Bipasha Pathirana, Pubudu N. Horne, Malcolm Power, Laura Szmulewicz, David Deakin Univ Sch Engn Networked & Sensing Control NSC Lab Waurn Ponds Vic Australia Florey Inst Neurosci & Mental Hlth Parkville Vic Australia Royal Victorian Eye & Ear Hosp RVEEH Balance Disorders & Ataxia Serv St Andrews Pl East Melbourne Vic Australia Alfred Hosp Cerebellar Ataxia Clin Praran Vic Australia

ISBN: (纸本)9781538613115

Cerebellar Ataxia (CA) is a neurological condition that leads to uncoordinated muscle movements, even affecting the production of speech. Effective biomarkers are necessary to produce an objective decision-making support tool for early diagnosis of CA in non-clinical environments. This paper investigates the reliability and effectiveness of vocal tract acoustic biomarkers for assessing CA speech. These features were tested on a database consisting of 52 clinically rated tongue-twister phrase 'British Constitution' and its 4 consonant-vowel (CV) excerpts /ti/, /ti/, /tion/ acquired from 30 ataxic patients and 22 healthy controls. Such a marker could be applied to objectively assess the severity of CA from a simple speaking test, contributing to the possibility of being translated into a computer based automatic module to screen the disease from the speech. All the vocal tract features explored in this study were statistically significant using Kolmogorov-Smirnov test at 5% level in distinguishing healthy and CA speech. Several machine learning classifiers with 5-fold cross-validations were implemented on the vocal features. It was observed that the intensity ratios corresponding to the 4 C-V excerpts in CA group showed an increased variability and produced the best classification accuracy of 84.6% using KNN classifier. Results motivate the use of vocal tract features for monitoring CA speech.

关键词： speech biomarker linear predictive coding spectrogram vocal tract measures acoustic analysis cerebellar ataxia

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ERROR RESILIENCE ENHANCEMENT FOR A ROBUST ADPCM AUDIO coding SCHEME

ERROR RESILIENCE ENHANCEMENT FOR A ROBUST ADPCM AUDIO CODING...

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

作者： Simkus, Gediminas Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781479928934

This work comprises an extension of a backward adaptive quantizer which is employed together with a robust lattice predictor in an ADPCM coding scheme. Predictors of the ADPCM audio coding schemes are often considered as the part most sensitive to transmission errors. Nevertheless, a single transmission error causes a short destabilization of the adaptive quantizer at the decoder side. Therefore, this destabilization boosts the deviation of the prediction filter at the decoder from the encoder side. Desired damping of the quantizer shortens synchronization periods. However, damping leads to degradation of the quantizer's adaptation properties and consequently to a decrease in audio quality. We show that the transmission of the quantizer's envelope to the decoder in short intervals and in combination with envelope error detector reduces the impairment of the reconstructed audio signal in noisy transmissions. An objective audio quality evaluation confirms significant quality improvement at BER higher than 10(-4) and no quality degradation if an ideal channel is employed.

关键词： Audio coding error robust transmission linear predictive coding robust ADPCM

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A Novel Method To Explore The Sparse Nature Of Prediction Parameters

A Novel Method To Explore The Sparse Nature Of Prediction Pa...

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International Conference on Control Communication & Computing India (ICCC)

作者： Sebastian, Mary Diana Varghese, Bibin Ajay, V. G. LBS Inst Technol Women Thiruvananthapuram Kerala India Indian Space Res Org VSSC Thiruvananthapuram Kerala India

ISBN: (纸本)9781467373494

Compressed sensing is a new paradigm to explore the sparse nature of the signals. Compressed sensing allows to acquire signals fundamentally below the uniform rate digitization followed by compression usually used for storage and transmission. linear predictive coding is the core of most of the speech coding algorithms. In this paper a novel method is presented to explore the sparse nature of the predictive parameters. Compressed sensing is applied on to the prediction parameters. Parameter estimation is posed as a 0-norm minimization problem. Alternative representations of the linear predictive parameters, i.e. the reflection coefficients and the line spectral frequencies are also considered and the performance is analysed using the spectral distortion measurement. The number of reflection coefficients and line spectral frequencies used for encoding can be reduced using compressed sensing.

关键词： Compressed sensing Speech coding linear predictive coding Line Spectral Frequencies Spectral Distortion Measurement

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THE ESTIMATION OF LINE SPECTRAL FREQUENCIES TRAJECTORIES BASED ON UNSCENTED KALMAN FILTERING

THE ESTIMATION OF LINE SPECTRAL FREQUENCIES TRAJECTORIES BAS...

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6th International Multi-Conference on Systems, Signals and Devices

作者： Boubakir, Chabane Berkani, Daoud Jijel Univ Dept Elect LAMEL Jijel Algeria Natl Polytechn Sch LSC Dept Elect Ei Harrach Algeria

ISBN: (纸本)9781424443451

In recent studies the Unscented Kalman Filter (UKF) was applied to some nonlinear systems. Several speech processing problems like the estimation of the formant trajectories, the state and parameter Kalman estimation for speech enhancement and the estimation of Line Spectral Frequency (LSF) trajectories. In this paper we apply the UKF to the estimation of LSF trajectories, in the case of synthetic and real noisy speech. The Expectation Maximization (EM) approach is used to iteratively estimate the LSF parameters. Furthermore, the Square-Root implementation of the UKF is used as it provides numeric stability and guarantees positive semi-definiteness of the state covariance.

关键词： Speech enhancement Kalman filtering Estimation linear predictive coding

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Study of Vocal Tract Shape Estimation Techniques for Children 7

Study of Vocal Tract Shape Estimation Techniques for Childre...

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7th International Conference on Communication, Computing and Virtualization (ICCCV)

作者： Veena, S. Wankhede, Nilashree S. Shah, Milind S. Fr C Rodrigues Inst Technol Dept Electon & Telecommun Engn Sect 9A Mumbai 400703 Maharashtra India

Accurate depiction of the vocal tract shape is essential for development of speech training aids for hearing impaired. Speech therapists have concluded from their research that if speech disorders are detected at an early stage then the rectification of those becomes faster and to a great accuracy. In case of adults, the vocal tract shape estimation can be performed by using techniques like X-Ray, MRI or using articulatory analysis by synthesis. Due to certain difficulties present in the methods mentioned like radiation effect and loud sound impulses produced by rapid switching of magnetic field are not considered suitable for children. Thus there is lack of availability of vocal tract shape data for children due to inability to use direct estimation methods for children. So it becomes necessary to carry out comparative study of vocal tract shape estimation techniques of children using indirect vocal tract shape estimation techniques. This paper aims to make available a considerable amount of data to researchers for study towards children's vocal tract shape using implemented techniques. The final objective of research work is to perform a detailed study on techniques available for vocal tract shape estimation, to implement and further compare the covariance and lattice methods for children using MATLAB software. Currently, the vocal tract shape estimation based on autocorrelation method is implemented and the results obtained are presented. (C) 2016 The Authors. Published by Elsevier B.V.

关键词： Vocal tract shape estimation linear predictive coding Speech training hearing impaired

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Arabic Speaker Identification System using Combination of DWT and LPC Features

Arabic Speaker Identification System using Combination of DW...

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International Conference on Open Source Systems and Technologies (ICOSST)

作者： Shah, Shahid Munir Ahsan, Syed Nadeem Iqra Univ Fac Engn Sci & Technol Dept Telecommun Engn Karachi Pakistan

ISBN: (纸本)9781479920549

speaker recognition plays a significant role in the field of human computer interaction. In the recent years, several researchers have contributed in this field and have successfully build machine learning models for automatic speaker recognition systems. In this paper, we propose an automatic speaker identification system for qaries (Quran reciter) of Arabic Language. For feature extraction discrete Wavelet Transform (DWT) and linear predictive coding (LPC) feature extraction techniques were used. Classification was performed by Random Forest (RF). In order to improve the identification accuracy DWT and LPC features were used singly (One at a time) and combined to train RF. Our system showed the best performance when RF was trained with the combination of features. In this case 90.90% recognition accuracy was achieved.

关键词： Speaker Recognition Discrete wavelet Transform linear predictive coding Random Forest

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Implementation of a Two-Channel LPC Vocoder

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IEEE Transactions on Communications 1983年第7期31卷 907-916页

作者： Shin, Byoung Cheol Lee, Hyeong Ho Un, Chong Kwan Lee, Sang Zee Song, Kil Ho Department of Electrical Engineering Korea Advanced Institute of Science and Technology Chongyangni Seoul South Korea Agency for Defense Development Daejon South Korea Gold Star Electric Company Osan South Korea

In this paper we describe hardware implementation of a 2400 bit/s two-channel linear predictive vocoder. The vocoder hardware uses the latest 2900 series “bit slice” microprocessor chips, and 2K RAM's and 1K ROM's of data memory. The system design is a two bus structure with a 208 ns cycle time. A significant feature of the vocoder is that it is capable of processing two voice channels simultaneously. Throughout the paper, emphasis is placed on details of firmware development of the vocoder system. Efficient design of the vocoder hardware is also discussed. © 1983 IEEE

关键词： linear predictive coding Vocoders Hardware Signal processing algorithms Microprogramming Microprocessor chips Narrowband Costs Speech analysis Random access memory

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