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检索条件"主题词=Linear Predictive coding"

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An alaryngeal speech enhancement method based on ADPCM approach

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

引用

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 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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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...

引用

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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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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Single-microphone wind noise reduction by adaptive postfiltering

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

引用

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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A 450bps Speech coding Algorithm Based on Multi-Mode Matrix Quantization

A 450bps Speech Coding Algorithm Based on Multi-Mode Matrix ...

引用

Congress on Image and Signal Processing, CISP

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

A 450 bps speech coder based on multi-frame structure and multi-mode matrix quantization is presented. The multiframe structure consisting of four 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 Bit rate linear predictive coding Autocorrelation Low pass filters Frequency Programmable logic arrays Delay Vocoders

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Partitioned Feature-based Classifier model

Partitioned Feature-based Classifier model

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Dong-Chul Park Department of Information Engineering Myongji University Yongin si South Korea

The Partitioned Feature-based Classifier (PFC) is proposed in this paper. PFC does not use entire feature vectors extracted from the original data at once to classify each datum, but use only groups of features related to each feature vector to classify data separately. In the training stage, the contribution rate calculated from each feature vector group is drawn throughout the accuracy of each feature vector group and then, in the testing stage, the final classification result is obtained by applying weights corresponding to the contribution rate of each feature vector group. The proposed PFC algorithm is applied to two audio data classification problems, a speech/music data classification problem and a music genre classification problem. The results demonstrate that conventional clustering algorithms can improve their classification accuracy when the proposed PFC model is used with them.

关键词： Feature extraction Music Data mining Multiple signal classification Speech Discrete wavelet transforms linear predictive coding Cepstrum Clustering algorithms Brightness

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An auditory BCI using voluntary mental response

An auditory BCI using voluntary mental response

引用

International IEEE/EMBS Conference on Neural Engineering, CNE

作者： Jing Guo Bo Hong Fei Guo Xiaorong Gao Shangkai Gao Department of Biomedical Engineering School of Medicine Tsinghua University Beijing China

A novel brain computer interface (BCI) is implemented in this study, which only depends on auditory modality. The subjects voluntary recognition of the property (e.g. voice laterality) of a target human voice makes the discriminability between brain responses to target and non-target voices in a random sequence. EEG data from eight subjects showed that the amplitude of N2 and late positive component (LPC) elicited by target voice was significantly higher than that by non-target ones. The area of N2 and LPC component was used as features for identifying the target among digit numbers 1-8, achieving an average accuracy of 89%.

关键词： Electroencephalography Electrodes linear predictive coding Human voice Auditory system USA Councils Brain computer interfaces Testing Neural engineering Enterprise resource planning

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A Research on Recognition of Tibetan Speakers Based on MFCC and Delta Features

A Research on Recognition of Tibetan Speakers Based on MFCC ...

引用

International Forum on Computer Science-Technology and Applications (IFCSTA)

作者： Yu Hongzhi China Minorities Information Technology Institute Northwest University for Nationalities Lanzhou Gansu China

Through experiment, the paper analyses and extracts MFCC and delta features of different Tibetan speakers and makes comparison. It shows that MFCC has certain stability between frames of the same speaker, while delta ... 详细信息

关键词： Mel frequency cepstral coefficient Speaker recognition Loudspeakers Feature extraction Speech recognition linear predictive coding Ear Data mining Character recognition Cepstral analysis

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