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检索条件"主题词=linear predictive coding"

共 2458 条记录，以下是281-290 订阅

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Optimization-quantization for least squares estimates and its application for lossless audio compression

Optimization-quantization for least squares estimates and it...

引用

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

作者： Florin Ghido Ioan Tabus Institute of Signal Processing Tampere University of Technology Finland

In this paper we study the problem of optimally quantizing the least square estimates and we introduce a method where the quantized estimate vector is obtained by a sequence of interleaved optimization-quantization scalar stages. We show how the general approach can be reduced to a simple and efficient algorithm when connecting it to the LDL T solver for general LS problems. The application to quantization of the linear prediction coefficients for audio lossless coding reveals the high performance of the approach, leading to topmost performance in the class of frame based audio coders, surpassing significantly the performance of the current MPEG4-ALS standard.

关键词： Least squares approximation Audio compression Quantization Vectors Performance loss linear predictive coding Optimization methods Transform coding Code standards Least squares methods

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Distinguishing depression and suicidal risk in men using GMM based frequency contents of affective vocal tract response

Distinguishing depression and suicidal risk in men using GMM...

引用

International Conference on Control, Automation and Systems ( ICCAS)

作者： Thaweesak Yingthawornsuk Richard G. Shiavi King Mongkut's University of Technology Thonburi Bangkok Thailand Department of Electrical Engineering and Computer Science Vanderbilt University Nashville TN USA

Two types of speech recording collected from three groups of male subjects clinically diagnosed with depression, remission from depression, and suicidal potential were analyzed and investigated for their acoustic features derived from sub-band energy over 0-2 KHz and GMM-based spectrum of the vocal tract response. Spontaneous and text-reading speech samples characterized by different vocal features revealed significant between-class separation power. Especially, features extracted from the reading speech seemed to provide more separability between classes than those of the spontaneous speech. Additionally, high classification accuracy confirmed that the studied features were capable of distinguishing groups of different diagnostic subjects efficiently. In classifying depressed/suicidal subjects the correct score of classification was at 88.5% for features extracted from reading speech samples, while 85.58% was found from classifying spontaneous speech features. These results were considered to be fairly high in classification performance, which is supportive of the promising ability to distinguish two diagnostic groups whose speech samples changed in their acoustic properties and correlated of serious mental states, known as vocal affects. Our findings suggested some clues in diagnosis of psychiatric disorders for psychiatrist.

关键词： Speech analysis Psychology linear predictive coding Feature extraction Frequency estimation Loudspeakers Control systems Automation Biomedical engineering Risk analysis

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Online programmable wireless sensor node for testing purpose

Online programmable wireless sensor node for testing purpose

引用

International Symposium on Electronics in Marine (ELMAR)

作者： Marko Malajner Karl Benkic Zarko Cucej FERI University of Maribor Maribor Slovenia FERI University of Maribor Maribor

ISBN: (纸本)9781424433643

This article describes the development of an online programmable wireless sensor node. The motivation is to develop a sensor node whose firmware can be upgraded online in wireless sensor network, i.e. the sensor node can be reprogrammed, when it is already deployed in the network. Secondly, we present our hardware design for a sensor node and how online programming.

关键词： Wireless sensor networks Testing Microcontrollers Microprogramming linear predictive coding Sensor phenomena and characterization Hardware Protocols Temperature sensors Sensor systems

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Speech Enhancement Using Spatial Processing and Modified Excitation Source for Underwater Speech Communication

Speech Enhancement Using Spatial Processing and Modified Exc...

引用

IEEE/IFIP International Conference on Embedded and Ubiquitous Computing, EUC

作者： Hyung-jun Ju Se-young Kim Jung-woo Han Ki-man Kim Seok-yeb Kang Department of Radio Communication Engineering Korea Maritime University Busan South Korea

In this paper, we propose new dereverberation method for underwater speech communication. The speech signal generated by a fixed one source. Then the signal received on 9 horizontal sensor array in shallow water environment. Finally, received signals are processed dereverberation. The received signals achieved delay-and-sum beamforming and then, signals calculates time delay both each signals. Next, we calculate Hilbert envelope using time delay information and LP residuals. At last, we acquire new LP residual combination using various parameters. Proposed technique shows better performance than previous dereverberation technique.

关键词： Speech enhancement Oral communication Delay effects Array signal processing Additive noise linear predictive coding Sensor arrays Microphones Log periodic antennas Underwater acoustics

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Fast Dereverberation for Hands-Free Speech Recognition

Fast Dereverberation for Hands-Free Speech Recognition

引用

Hands-Free Speech Communication and Microphone Arrays, HSCMA

作者： Randy Gomez Jani Even Hiroshi Saruwatari Kiyohiro Shikano ACCMS Kyoto University Kyoto Japan Graduate School of Information Science NAIST Japan

A robust dereverberation technique for real-time hands-free speech recognition application is proposed. Real-time implementation is made possible by avoiding time-consuming blind estimation. Instead, we use the impulse response by effectively identifying the late reflection components of it. Using this information, together with the concept of Spectral Subtraction (SS), we were able to remove the effects of the late reflection of the reverberant signal. After dereverberation, only the effects of the early component is left and used as input to the recognizer. In this method, multi-band SS is used in order to compensate for the error arising from approximation. We also introduced a training strategy to optimize the values of the multi-band coefficients to minimize the error.

关键词： Speech recognition Reflection Reverberation linear predictive coding Robustness Microphone arrays Hidden Markov models Information science Automatic speech recognition Signal processing

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Efficient production technology for microcrystalline silicon solar cells using a Localized Plasma Confinement (LPC) CVD method

Efficient production technology for microcrystalline silicon...

引用

2008 33rd IEEE Photovoltaic Specialists Conference

作者： Toshie Kunii Kazuya Murata Mitsuhiro Matsumoto Kunihiro Kawamoto Yasutaka Kobayashi Youichirou Aya Makoto Nakagawa Akira Terakawa Makoto Tanaka Advanced Energy Research Center SANYO Electric Company Limited Kobe Hyogo Japan

A very high deposition rate for high-quality microcrystalline silicon (μc-Si) films has been achieved under very high-pressure conditions (≫ 1,000 Pa) using a Localized Plasma Confinement (LPC)-CVD method which has a special cathode. The uniformity of the μc-Si film thickness was 2.4% on a 55×65 cm 2 glass substrate with a deposition rate of 2.7 nm/s. We also achieved maximum conversion efficiency of 11.4% for an a-Si/μc-Si tandem solar cell (1 cm 2 ) on a 20×20 cm 2 glass substrate and average conversion efficiency of 9.84% for a-Si/μc-Si tandem solar cells (1 cm 2 ) on a 55×65 cm 2 glass substrate with a deposition rate (Rd) of 1.8 nm/s. These results indicate that LPC-CVD method is a good candidate as an effective production technology for large-area, high-performance μc-Si thin-film solar cells.

关键词： Production Silicon Photovoltaic cells Plasma confinement linear predictive coding Glass Substrates Semiconductor films Cathodes Transistors

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Voice Activity Detection Using Entropy in Spectrum Domain

Voice Activity Detection Using Entropy in Spectrum Domain

引用

Australasian Telecommunication Networks and Applications Conference, ATNAC

作者： Meysam Asgari Abolghasem Sayadian Mohsen Farhadloo Elahe abouie Mehrizi Department of Electrical Engineering Amirkabir University of Technology슠 Tehran Iran Department of Biomedical and Computer Engineering University of Binningham Birmingham UK

In this paper we develop a voice activity detection algorithm based on entropy estimation of magnitude spectrum. In addition, the likelihood ratio test (LRT) is employed to determine a threshold to separate of speech segments from non-speech segments. The distributions of entropy magnitude of clean speech and noise signal are assumed to be Gaussian. The application of the concept of entropy to the speech detection problem is based on the assumption that the signal spectrum is more organized during speech segments than during noise segments. One of the main advantages of this method is that it is not very sensitive to the changes of noise level. Our simulation results show that the entropy based VAD is high performance in low signal to noise ratio (SNR) conditions (SNR < 0 dB).

关键词： Entropy Speech enhancement Speech coding Testing Light rail systems Signal to noise ratio Bandwidth Bit rate Radio frequency linear predictive coding

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Doppler-variant modeling of the vocal tract

Doppler-variant modeling of the vocal tract

引用

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

作者： Axel Heim Uli Sorger Florian Hug Institute of Telecommunications and Applied Information Theory Ulm University Germany Computer Science and Communications University of Luxembourg Luxembourg

A common technique to deploy linear prediction to non-stationary signals is time segmentation and local analysis. Variations of a process within such a segment cause inaccuracies. In this paper, we model the temporal changes of linear prediction coefficients (LPCs) as a Fourier series. We obtain a compact description of the vocal tract model limited by the predictor order and the maximum Doppler frequency. Filter stability is guaranteed by all-pass filtering, deploying the human ear's insensitivity to absolute phase. The periodicity constraint induced by the Fourier series is counteracted by oversampling in the Doppler domain. With this approach, the number of coefficients required for the vocal tract modeling is significantly reduced compared to a LPC system with block-wise adaptation while exceeding its prediction gain. As a by-product it is found that the Doppler frequency of the vocal tract is in the order of 10 Hz. A generalization of the algorithm to an auto-regressive moving average model with time-correlated filter coefficients is straight forward.

关键词： linear predictive coding predictive models Fourier series Filters Frequency Speech coding Speech processing Shape Entropy Information theory

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Age Bracket Judgment by LPC Spectrum Inclination of Voice: Comparing with Detection of Pitch Alternation / Formant Q in the Vowel Term

Age Bracket Judgment by LPC Spectrum Inclination of Voice: C...

引用

International Symposium on Knowledge Acquisition and Modeling, KAM

作者： Kazuhide Okada Vehicle Control System Development Department Higashi-Fuji Technical Center Toyota Motor Corporation Susono Shizuoka Japan

If it is known the said person is the young or old, when we make persons utter the same vocabulary, it is possible to guide only old man kindly and repeatedly by speech synthesis and otherwise to give only young generation the cautious voice toward their own reckless action by mosquito sound after its detection as application. We observed the alternation of the pitch, Formant Q which means power steepness at formant frequency on Spectrum, and inclination of LPC Spectrum in order to realize its *** its result we could know the inclination of LPC Spectrum of young generation becomes steeper toward the low frequency compared with that of old one clearly, in spite that it is uncertain that change of the number of the vibration at the vocal cords is big or small and steep or gentle power mountain exist at the 1st formant frequency in correspondence with the generation. In fact when the inclination of LPC Spectrum is used as feature parameter, the average recognition ratio toward an age bracket was 88.8%, besides when change of pitch or Formant Q was used, its ratio is under 60%.

关键词： linear predictive coding Frequency Face detection Vehicle detection Power generation Pediatrics Knowledge acquisition Vehicles Control system synthesis Electronic mail

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Target Feature Extraction for Passive Sonar Based on Two Cepstrums

Target Feature Extraction for Passive Sonar Based on Two Cep...

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International Conference on Bioinformatics and Biomedical Engineering (ICBBE)

作者： Geming Liu Chao Sun Yixin Yang Engineering Acoustics Institute Northwestern Polytechnical University Xi'an China

A passive sonar target can be regarded as a sounder from the viewpoint of sonar operators. The impulse response of the sounder in the cepstrum domain, which shows physical features of targets, can be obtained from the radiated-noise of targets using LPC cepstrum and Mel cepstrum. A set of cepstrum-domain features is extracted based on the above two cepstrums of the impulse response. The neural network target classifier was designed using cepstrum-domain features. The classification experiments were carried out for three different kinds of targets based on practical data. The experimental results show that the feature extraction method based on two cepstrums are useful.

关键词： Feature extraction Sonar Cepstrum Acoustic noise Time domain analysis Target recognition linear predictive coding Noise shaping Acoustical engineering Data mining

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