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检索条件"主题词=Linear predictive coding"

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A two pass classifier for utterance rejection in keyword spotting

A two pass classifier for utterance rejection in keyword spo...

引用

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

作者： R.A. Sukkar J.G. Wilpon AT and T Bell Laboratories Inc. Naperville IL USA

A classifier for utterance rejection in a hidden Markov model (HMM) based speech recognizer is presented. This classifier, termed the two-pass classifier, is a postprocessor to the HMM recognizer, and consists of a two-stage discriminant analysis. The first stage employs the generalized probabilistic descent (GPD) discriminative training framework, while the second stage performs linear discrimination combining the output of the first stage with HMM likelihood scores. In this fashion the classification power of the HMM is combined with that of the GPD stage which is specifically designed for keyword/nonkeyword classification. Experimental results show that, on two separate databases, the two-pass classifier significantly outperforms a single-pass classifier based solely on the HMM likelihood scores.< >

关键词： Hidden Markov models Cepstral analysis Monitoring Automatic speech recognition Isolation technology Speech recognition Databases Telephony Robustness linear predictive coding

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The application of a functional perceptual model of speech to variable-rate LPC systems

The application of a functional perceptual model of speech t...

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

作者： R. Viswanathan J. Makhoul R. Wicke Bolt Beranek and Newman 슠Inc. Cambridge MA USA

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Comparisons of maximum system lifetime in diverse scenarios for body sensor networks

Comparisons of maximum system lifetime in diverse scenarios ...

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International Conference on Computer and Information Technology (ICCIT)

作者： Nazrul Islam Mondal Shahid Uz Zaman Abdullah Al Masud Jahangir Alam Department of Computer Science & Engineering Rajshahi University of Engineering & Technology Bangladesh Department of Electronic and Telecommunication International Islamic University Chittagong Bangladesh Department of Computer Science & Engineering Rajshahi University of Engineering and Technology Bangladesh Department of Electronic and Telecommunication International Islamic University Chittagong Bangladesh

In this paper, the main target is to compare the system lifetimes in diverse scenarios based on our new energy dissipation model in the body sensor network as well as maximize the system lifetime of sensor nodes when they will make communication among body sensors and personal communication unit. Nowadays, the ultra low energy consumption is the very much important challenge for medical applications. The best compression technique like LPC is selected for energy saving based on some calculations. This research work studies and analyzes the transceiver energy consumption for different compression algorithms and select the best technique like LPC for energy saving. Formulation of a linear programming problem is also the important part of this research work, where is to maximize the system lifetime which is equivalent to the time until the first node runs out of battery. Maximum system lifetimes are calculated by MATLAB optimization technique using and without using efficient compression algorithm like LPC in various environments. Results show that maximum system lifetimes calculated in different scenarios using efficient compression technique like LPC is better than without using compression technique.

关键词： Body sensor networks linear predictive coding Sensor systems Energy consumption Compression algorithms Energy dissipation Mathematical model Medical services Biomedical equipment Algorithm design and analysis

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Promising concepts and technologies for future power delivery systems

Promising concepts and technologies for future power deliver...

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International Universities Power Engineering Conference (UPEC)

作者： Phuong H. Nguyen Wil L. Kling Johanna M. A. Myrzik Eindhovan University of Technology Netherlands Delft University of Technnology Netherlands

The transition from a conventional delivery system to a suitable robust distribution system is emerging to adapt with different possible scenarios of future development. This paper gives a particular vision of future grid with its main requirements. An investigation of suitable concepts and technologies which draw out attentions at the present has been carried out. They are discussed regarding mentioned requirements of sustainability, efficiency, flexibility and intelligence. Active network is then introduced as the backbone of the future power delivery system. Besides, multi- agent system (MAS) is described as a potential technology to cope with anticipated challenges of future grid operation. The research described is under the framework of the Electricity Infrastructure of the Future (EIT) project carried out by cooperation of TU/e, KEMA and ECN (***).

关键词： Distributed control Supply and demand Robustness Multiagent systems Control systems Voltage control Power system stability Communication system control linear predictive coding Spine

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Automatic speech recognition using hidden Markov models and artificial neural networks

Automatic speech recognition using hidden Markov models and ...

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International Conference on Neural Networks

作者： N.M. Botros M. Siddiqi M.Z. Deiri Department of Electrical Engineering Southem Illinois University Carbondale IL USA

An algorithm is presented for isolated-word recognition, taking into consideration the duration variability of the different utterances of the same word. The algorithm is based on extracting acoustical features from the speech signal and using them as the input to multilayer perceptrons neural networks. The backpropagation algorithm is used to train the networks. The hidden Markov model (HMM) is implemented to extract temporal features (states) from the speech signal. The input vector to the network consists of 16 cepstral coefficients, two delta cepstral coefficients, and five elements to represent the state. The networks are trained to recognize the correct words and to reject the wrong words. The training set consists of ten words (digit zero to digit nine), each uttered seven times, by three different speakers. The test set consists of three utterances of each of the ten words. The authors' results show the ability to recognize all of these words.< >

关键词： Automatic speech recognition Hidden Markov models Artificial neural networks Cepstral analysis Feature extraction Band pass filters Vectors linear predictive coding Multilayer perceptrons Neural networks

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Spectral stability based event localizing temporal decomposition

Spectral stability based event localizing temporal decomposi...

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

作者： A.C.R. Nandasena M. Akagi Graduate School of Information Science Japan Advanced Institute of Science and Technology Japan

A new approach to temporal decomposition (TD) of speech, called "spectral stability based event localizing temporal decomposition", abbreviated S/sup 2/ BEL-TD, is presented. The original method of TD proposed by Atal (1983) is known to have the drawbacks of high computational cost, and the instability of the number and locations of events. In S/sup 2/ BEL-TD, the event localization is performed based on a maximum spectral stability criterion. This overcomes the instability problem of events of the Atal's method. Also, S/sup 2/ BEL-TD avoids the use of the computationally costly singular value decomposition routine used in the Atal's method, thus resulting in a computationally simpler algorithm of TD. Simulation results show that an average spectral distortion of about 1.5 dB can be achieved with LSF as the spectral parameter. Also, we have shown that the temporal pattern of the speech excitation parameters can also be well described using the S/sup 2/ BEL-TD technique.

关键词： Speech Singular value decomposition Information science Costs Computational modeling linear predictive coding Solids Stability criteria

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Automatic classification of singing voice quality

Automatic classification of singing voice quality

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International Conference on Intelligent Systems Design and Applications (ISDA)

作者： B. Kostek P. Zwan Multimedia Systems Department Gdańsk University of Technology Poland

In the paper problems related to the classification of singing voice quality are presented. For this purpose a database consisting of singers' sample recordings is constructed and parameters are extracted from recorded voice of trained and untrained singers. The parameterization process is based on both voice source and formant analysis of a singing voice. These parameters are explained as to their physical interpretation and analyzed statistically in order to diminish their number. The statistical analysis is based on the Fisher statistic. In such a way a feature vector of a singing voice is formed. Decision systems based on neutral networks and rough sets are utilized in the context of the voice type and voice quality classification. Results obtained in the automatic classification performed by both decision systems are compared. A possibility to classify automatically type/quality of voice is judged. The methodology proposed provides means for discerning trained and untrained singers.

关键词： Frequency Statistical analysis Speech analysis Rough sets Biomechanics Intelligent systems Signal analysis linear predictive coding Paper technology Multimedia systems

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Voice over IP with JVOIPLIB and JRTPLIB

Voice over IP with JVOIPLIB and JRTPLIB

引用

Conference on Local Computer Networks (LCN)

作者： J. Liesenborgs W. Lamotte F. Van Reeth Expertisecentrum Digitale Media Limburgs Universitair Centrum Diepenbeek Belgium

ISBN: (纸本)0769513212

Voice over IP (VoIP) can be used in a wide variety of applications, all having different requirements. We present JVOIPLIB and JRTPLIB, a VoIP library and an RTP library respectively. Together they make it possible to easily add VoIP to various types of applications. Both libraries are written in an object-oriented style in C++, are open-source and are both very extensible. Several measures have been taken to allow good synchronization between the communicating parties.

关键词： Internet telephony Virtual environment Delay linear predictive coding Software libraries Open source software Speech processing Signal processing Jitter Degradation

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Classification Using the Local Probabilistic Centers of k-Nearest Neighbors

Classification Using the Local Probabilistic Centers of k-Ne...

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International Conference on Pattern Recognition

作者： Bo Yu Li Yun Wen Chen Department of Computer Science and Engineering Fudan University Shanghai China

ISBN: (纸本)0769525210

In high dimensional feature space with finite samples, severe bias can be introduced in the nearest neighbor algorithm. In this paper, we propose a new classification method, which performs classification task based on local probability center of each class. Moreover, this prototype-based method classifies the query sample by using two measures, one is the distance between query and local probability centers, the other is the posterior probability of query. Although both measures are effect, the experiments show the second one is the better. The investigation results prove that this method improves the classification performance of nearest neighbor algorithm substantially

关键词： Nearest neighbor searches Neural networks Support vector machines linear predictive coding Prototypes Computational efficiency Computer science Pattern classification Training data Support vector machine classification

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A comparison of SPOX and C implementations of a LPC vocoder on the TMS320C30 EVM

A comparison of SPOX and C implementations of a LPC vocoder ...

引用

IEEE Southeastcon

作者： D.A. Kilpatrick T.C. Jannett Department of Electrical and Computer Engineering University of Alabama Birmingham Birmingham AL USA

The paper describes a comparison of a C implementation of a linear predictive voice coder (LPC) and an implementation based on Spectron Microsystem's Signal Processing Operating System (SPOX). The hardware platform was a Texas Instruments TMS320C30 Evaluation Module. The SPOX and C implementations were compared based on execution time, ease of program development and maintenance, and portability to different hardware platforms. The vocoder algorithms and the results of the comparison of both implementations are presented.

关键词： linear predictive coding Vocoders Speech synthesis Speech analysis Filters Hardware Signal synthesis Synthesizers Humans Signal processing algorithms

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