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检索条件"主题词=linear predictive coding"

共 2456 条记录，以下是211-220 订阅

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Normal probability testing of snore signals for diagnosis of obstructive sleep apnea

Normal probability testing of snore signals for diagnosis of...

引用

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

作者： H. Ghaemmaghami U. R. Abeyratne C. Hukins School of Information Technology and Electrical Engineering University of Queensland Brisbane Australia Sleep Diagnostic Unit Princess Alexandra Hospital Brisbane Australia

Obstructive sleep apnea (OSA) is a highly prevalent disease in which upper airways are collapsed during sleep, leading to serious consequences. The standard method of OSA diagnosis is known as polysomnography (PSG), which requires an overnight stay in a specifically equipped facility, connected to over 15 channels of measurements. PSG requires (i) contact instrumentation and, (ii) the expert human scoring of a vast amount of data based on subjective criteria. PSG is expensive, time consuming and is difficult to use in community screening or pediatric assessment. Snoring is the most common symptom of OSA. Despite the vast potential, however, it is not currently used in the clinical diagnosis of OSA. In this paper, we propose a novel method of snore signal analysis for the diagnosis of OSA. The method is based on a novel feature that quantifies the non-Gaussianity of individual episodes of snoring. The proposed method was evaluated using overnight clinical snore sound recordings of 86 subjects. The recordings were made concurrently with routine PSG, which was used to establish the ground truth via standard clinical diagnostic procedures. The results indicated that the developed method has a detectability accuracy of 97.34%.

关键词： Sleep apnea Testing Australia Instruments Humans Information technology linear predictive coding Databases USA Councils Diseases

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Speech Emotion Recognition Research Based on the Stacked Generalization Ensemble Neural Network for Robot Pet

Speech Emotion Recognition Research Based on the Stacked Gen...

引用

Chinese Conference on Pattern Recognition (CCPR)

作者： Yongming Huang Guobao Zhang Xiaoli Xu School of Automation South-East University Nanjing Jiangsu China

In this paper, we present an emotion recognition system using the stacked generalization ensemble neural network for special human affective state in the speech signal. 450 short emotional sentences with different contents from 3 speakers were collected as experiment materials. The features relevant with energy, speech rate, pitch and formant are extracted from speech signals. Stacked generalization ensemble neural networks are used as the classifier for 5 emotions including anger, calmness, happiness, sadness and boredom. First, compared with the traditional BP network or wavelet neural network, the results of experiments show that the stacked generalization ensemble neural network has faster convergence speed and higher recognition rate. Second, after discussing the advantage and disadvantage between different ensemble neural networks, suitable decision will be made for robot pet.

关键词： Emotion recognition Neural networks Robots Positron emission tomography Humans Speech processing Filtering linear predictive coding Frequency Robotics and automation

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A method to determine the distributed control setting of looping devices for active distribution systems

A method to determine the distributed control setting of loo...

引用

Power Tech Conference

作者： N. Okada System Engineering Research Laboratory Central Research Institute of Electric Power Industry Komae Tokyo Japan

To improve operational flexibility, we proposed the loop or mesh structure for distribution systems using looping device called loop power flow controller (LPC) at the opened switch which can connect the adjoining feeder as an active distribution system. LPC is able to optimally control distribution systems in the points of reducing voltage rise, voltage fluctuation and loss minimum power flow control, and so on, with acquisition of state of distribution systems. Adapting to the area that cannot use communication system, points of view at the power quality for the transition phenomena, as assistance to the central control, we also required the distributed control using local voltage information for the LPC control. The purposes of this paper are the application of the proposed determination method for control coefficients using optimal operation pattern of LPC and local voltage information to two LPC and the confirmation of its validity by simulation. The distributed control using these control coefficients was verified by two 6.6 kV 100 kVA BTB (back to back) type LPC and distribution system testing facility in CRIEPI.

关键词： Distributed control linear predictive coding Control systems Voltage control Communication system control Open loop systems Switches Optimal control Voltage fluctuations Centralized control

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The computation of line spectrum pair frequencies using Tschirnhaus transform

The computation of line spectrum pair frequencies using Tsch...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Shi-Huang Chen Yaotsu Chang Chang Jian Yu Syuan Department of Computer Science & Information Engineering Shu Te University Kaohsiung Taiwan Department of Applied Mathematics and the Department of Information Engineering I-Shou University Kaohsiung Taiwan Department of Applied Mathematics National Sun Yat-Sen University Kaohsiung Taiwan

ISBN: (纸本)9781424438273

In this paper, a new algorithm based on the Tschirnhaus transforms is developed to reduce the computation complexity of the 10-order line spectrum pairs (LSP) frequencies. The first step of the proposed algorithm is to derive a quartic equation from the 1st derivative of the given 5-degree LSP polynomial. Then the extremes of the 5-degree LSP polynomial can be found by applying the Tschirnhaus transform to the above quartic equation. By the use of these extremes as the initial approximations, one can easily solve the roots of the 5-degree LSP polynomial via the Newton's method and get the accurate LSP frequencies. One of the main advantages of the proposed algorithm is the rapid root determination of a quartic equation without complex number operations and resulting in considerable computational saving. Compared to other methods, the proposed algorithm can determine the precise LSP frequencies with the lowest computational complexity.

关键词： Frequency Polynomials linear predictive coding Equations Mathematics Computational complexity Discrete cosine transforms Computer science Cities and towns Newton method

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An alaryngeal speech enhancement method based on ADPCM approach

An alaryngeal speech enhancement method based on ADPCM appro...

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Agustin Razo-Chavez Mariko Nakano-Miyatake Hector Perez-Meana Mechanical and Electrical Engineering School National Polytechnic Institute Mexicali Mexico

An esophageal speech enhancement system is proposed to improve the intelligibility and quality of esophageal speech. Proposed system identifies the voiced segments of esophageal speech signal, by using pattern recognition methods, and replaces these by their equivalent voiced segments of normal speech, while keeps the silence and unvoiced segments without change. Evaluation results, using objective and subjective evaluation methods, show that proposed system provides a fairly good improvement of the quality and intelligibility of, esophageal speech signals.

关键词： Speech enhancement Esophagus Speech synthesis Frequency estimation Speech analysis Larynx Signal restoration Signal synthesis Filters linear predictive coding

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A Modified Adaptive Algorithm for Formant Bandwidth in Whisper Conversion

A Modified Adaptive Algorithm for Formant Bandwidth in Whisp...

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International Asia Conference on Informatics in Control, Automation and Robotics, CAR

作者： Gang Lv Heming Zhao School of Electronic Information Soochow University Suzhou China

The whisper conversion technology is to transform undistinguished whispers with lower SNR into clear normal speech, and it has important application prospect in mobile communication. Because the whisper speech is stirred by the yawp source, its formant position shifts and its bandwidth increases, which induces the problem of formant combination occurs in the whisper conversion. By analyzing the power spectrum, in this article, we proposed a modified adaptive algorithm for formant bandwidth. Based on the rule that the pole power does not change, the algorithm has resolved the problem of formant combination by modifying the formant bandwidth of whisper before implementing formant conversion. The experimental results with six Chinese Mandarin monophthong phoneme conversions proved the validity of the algorithm.

关键词： Adaptive algorithm Bandwidth Mobile communication Frequency conversion Robotics and automation Algorithm design and analysis Speech analysis linear predictive coding Asia Informatics

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A 300bps speech coding algorithm based on multi-mode matrix quantization

A 300bps speech coding algorithm based on multi-mode matrix ...

引用

International Conference on Wireless Communications and Signal Processing (WCSP)

作者： Xia Zou Xiongwei Zhang Yafei Zhang People''s Liberation Army University of Science and Technology Nanjing China

A 300 bps speech coder based on multi-frame structure and multi-mode matrix quantization is presented. The multi-frame structure consisting of six frames is adopted to reduce the algorithm delay. The parameter matrices are classified into different modes based on the voicing vector information of superframe. To improve speech quality, a dynamic bit allocation scheme is developed. Experimental results show that the speech quality of the proposed vocoder is intelligible with good naturalness.

关键词： Speech coding Quantization linear predictive coding Bit rate Power harmonic filters Autocorrelation Low pass filters Frequency Decoding Random number generation

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Sigma-delta analog to LPC feature converters for portable recognition interfaces

Sigma-delta analog to LPC feature converters for portable re...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Shantanu Chakrabartty Amit Gore Department of Electrical and Computer Engineering Michigan State University East Lansing MI USA

ISBN: (纸本)9781424438273

For many recognition systems, the feature extraction unit forms the most computationally intensive and power consuming component. In this paper, we present a design of an analog-to-information converter that directly produces a pulse-encoded representation of linear predictive coded (LPC) features corresponding to an input analog signal. At the core of proposed design is a sigma-delta modulation procedure that is embedded within a learning step. Measured results from a fabricated prototype in a 0.5 mum CMOS technology demonstrate the real-time functionality of the learner in extracting 6-dimensional online LPC features from input speech signal while consuming only 450 muW.

关键词： Delta-sigma modulation linear predictive coding Feature extraction Speech recognition Analog computers Prototypes CMOS technology Circuit testing Computer interfaces Portable computers

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Single-microphone wind noise reduction by adaptive postfiltering

Single-microphone wind noise reduction by adaptive postfilte...

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IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： Elias Nemer Wilf Leblanc Broadcom Corporation Irvine CA USA

This paper presents a novel time-domain algorithm for detecting and attenuating the acoustic effect of wind noise in speech signals originating from mobile terminals. The detection part makes use of metrics that exploits the properties of the spectral envelop of wind noise as well as its non-periodic and non-harmonic nature. LPC analyses of various orders are carried out and the results used to distinguish between wind and speech frames and to estimate the magnitude and location of the wind noise `resonance'. The suppression part entails constructing a parameterized postfilter of an appropriate order having a `null' where the wind noise `resonance' is. Wind-only frames are used to estimate the wind noise energy, from which the emphasis parameters of the post-filter are adjusted to provide an appropriate attenuation. The proposed scheme may be combined with background-noise suppression algorithms, or with speech-formant-enhancing post-filters in the context of a speech codec.

关键词： Noise reduction Acoustic signal detection Acoustic noise Speech enhancement Speech analysis Resonance Time domain analysis linear predictive coding Wind energy Attenuation

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Audio-based classification of speaker characteristics

Audio-based classification of speaker characteristics

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Promiti Dutta Alexander Haubold Columbia University New York NY USA

The human voice is primarily a carrier of speech, but it also contains non-linguistic features unique to a speaker and indicative of various speaker demographics, e.g. gender, nativity, ethnicity. Such characteristics are helpful cues for audio/video search and retrieval. In this paper, we evaluate the effects of various low-, mid-, and high-level features for effective classification of speaker characteristics. Low-level signal-based features include MFCCs, LPCs, and six spectral features; mid-level statistical features model low-level features; and high-level semantic features are based on selected phonemes in addition to mid-level features. Our data set consists of approximately 76.4 hours of annotated audio with 2786 unique speaker segments used for classification. Quantitative evaluation of our method results in accuracy rates up to 98.6% on our test data for male/female classification using mid-level features and a linear kernel support vector machine. We determine that mid- and high-level features are optimal for identification of speaker characteristics.

关键词： linear predictive coding Mel frequency cepstral coefficient Speech analysis Aggregates Feature extraction Frequency estimation Testing Automatic speech recognition Indexing Covariance matrix

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