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检索条件"主题词=Linear Predictive coding"

共 2458 条记录，以下是361-370 订阅

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A 53% High Efficiency GaAs Vertically Integrated Multi-junction Laser Power Converter

A 53% High Efficiency GaAs Vertically Integrated Multi-junct...

引用

Device Research Conference

作者： D. Krut R. Sudharsanan T. Isshiki R. King N.H. Karam Boeing Company Spectrolab Inc. Sylmar CA USA

In this paper, a new vertical multi-junction 2 V laser power converter with an efficiency of over 52% is successfully and fabricated. This efficiency was obtained with high level of direct laser illumination at 810 nm.

ISBN: (纸本)9781424411016;1424411017

关键词： Gallium arsenide Power lasers Optical losses Photovoltaic systems Solar power generation linear predictive coding Current density Optical devices Optical design Fiber lasers

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linear Prediction Based Speech Codec on General Purpose Microcontroller

Linear Prediction Based Speech Codec on General Purpose Micr...

引用

Annual Siberian Russian Workshop on Electron Devices and Materials (EDM)

作者： Alexander A. Baulin CAD chair Siberian State University of Telecommunications and Informatics Novosibirsk Russia

This paper considers realization issues of linear prediction based speech codec on single-chip system with ARM7TDMI core.

关键词： Speech codecs Microcontrollers linear predictive coding Frequency Bandwidth Manufacturing Internet telephony Costs Informatics Radio control

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Subframe Interpolation Optimized coding of LSF Parameters

Subframe Interpolation Optimized Coding of LSF Parameters

引用

ACIS International Conference on Software Engineering, Artificial Intelligence, Networking, and Parallel/Distributed Computing (SNPD)

作者： Linliang Chen Yanping Yu Kuang Wang Department of Information Science Electronic Engineering University of Zhejiang Hangzhou China

In many speech coding systems, frames are further divided into subframes. The line spectral frequency (LSF) parameters of each subframe are used to construct the linear predictive coding (IPC) synthesis filter but only the LSF parameters of the last subframe in each frame are quantized and transmitted while those of others are obtained by interpolating in decoder. Conventional coding schemes are designed to try to minimize the spectral distortion of the quantized LSF vectors without considering those of interpolated ones. However, a considerable performance could be gained by exploiting this subframe interpolation operation. This paper presents a novel and efficient LSF coding approach. The proposed subframe- weighted LSF distance measure allows a coder to improve the LSF coding performance over all subframes of all frames. Simulation results show that both average spectral distortion (SD) and outliers of the proposed algorithm are improved. Therefore it is more excellent than conventional ones.

关键词： Interpolation Speech coding Frequency synthesizers linear predictive coding Speech synthesis Nonlinear filters Decoding Vectors Distortion measurement Weight measurement

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Measurement of Urdu Speaker's Speaking Speed in Ordinary Conversation

Measurement of Urdu Speaker's Speaking Speed in Ordinary Con...

引用

IEEE International Conference on Multi Topic

作者： J.S. Mirza H. bin Liaqat S.A. Hayat Department of Computer Science COMSATS Institute of Information technology Lahore Pakistan Department of Computer Science COMSATS Institute of Information technology Lahore.

ISBN: (纸本)1424415535;9781424415526

Speaker identification traditionally employs technologies of MRCC and LPC, and most of the proven existing technologies are strictly text-dependent. New technologies are text independent. Whatever technologies are used for speaker identification their performance degrades if the list of the speakers stored for identification grows. Therefore, if other pertinent parameters are added to the extracted data from the speaker's speech, speaker identification will get enhanced. This research aims to enhance speaker identification by adding a new parameter namely speaker's speaking speed to the extracted data to enhance the accuracy of the speaker identification. This paper has manually determined the speaking speed from many recorded conversations, which were saved in computer by Bluetooth. A reputed software Praat was used to watch sound graphics and to make calculations. The Praat's calculation confirm that our study can be used to make an algorithm to automatically determine correct speaker speed.

关键词： Velocity measurement Loudspeakers Hidden Markov models Degradation Data mining Speech enhancement Information technology linear predictive coding Bluetooth Watches

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Analysis of Urdu Syllables and Proposed Algorithm Structure for Determination of Speaker's Speaking Rate

Analysis of Urdu Syllables and Proposed Algorithm Structure ...

引用

IEEE International Conference on Multi Topic

作者： J S Mirza Hannan bin Liaqat S.A. Hayat Department of Computer Science COMSATS Institute of Information Technology Lahore Pakistan

ISBN: (纸本)1424415535;9781424415526

In this paper it is reported that definite relationships exist between the number of syllables in any Urdu text and the long vowels, short vowels and consonants these syllables contain. This information is useful in preparing an automatic algorithm to calculate speaking speed of a speaker. For every 100 syllables there are 38 short vowels, 58 long vowels and 3% of diphthongs. This percentage holds good for almost any number of syllables greater then about 50. The consonants have also been counted in syllables and found to be 135% of syllables. For every 100 syllables there are about 135 consonants, and this percentage holds true for syllables greater then about 50 syllables. These numbers help us calculate the probability of the long, short vowels and the consonants in any Urdu text, which can in turn be used in an algorithm to automatically calculate the speaking speed.

关键词： Algorithm design and analysis Information analysis Information technology Probability Speech Cepstral analysis Mel frequency cepstral coefficient linear predictive coding Robustness

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Measurement and Analysis of “Yes” and “No” Responses by Auditory Stimuli Questions in Human EEG

Measurement and Analysis of “Yes” and “No” Responses by ...

引用

Frontiers in the Convergence of Bioscience and Information Technologies, FBIT

作者： Ssanghee Seo Haifeng Chen Donghee Ye Jungtae Lee Kyungjae Ha Department of Computer Science & Engineering Pusan National University Busan South Korea Department of Computer Science & Engineering Kyungnam University Masan South Korea

Patients with severe physical disorders but who are cognitively intact find it difficult to communicate with the outside world. Researches on the brain-computer interface (BCI) are aimed to help such patients. This study analyzes the characteristics of brainwaves when a subject thinks of a "yes" or "no" response to an auditory stimuli question. Experiment using auditory stimuli of the form of questions is performed. We discriminate between the activities of the brainwaves of the "yes" and "no " responses using the alpha power. Nineteen healthy subjects (7 females and 12 males), aged between 20 and 35 years, volunteered to participate in the experiment. In the experiment, for 14 out of 19 subjects (73.68%), the average alpha power for the "yes" response is lower than that for the "no " response.

关键词： Humans Electroencephalography Enterprise resource planning Brain computer interfaces Frequency Computer science Resource management linear predictive coding Information technology Information analysis

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The Study of the Classification of Chinese Folk Songs by Regional Style

The Study of the Classification of Chinese Folk Songs by Reg...

引用

International Conference on Semantic Computing

作者： Yi Liu JiePing Xu Lei Wei Yun Tian Computer Science Department Information School RenMin University of China China

This paper discusses a method of studying the region style classification of Chinese folk songs with support vector machine (SVM). According to geographical region of China, We have classified Chinese folk songs into 10 major categories, and used 500 Chinese folk songs in our experiment. 74 features have been extracted from audio files of the songs, and classified by an audio classifier on SVM. The experiment results show that sampling rate is not directly proportional to classification accuracy; SVM without feature selection is a very effective classification method for region style classification; the combination of 13-dimension MFCC and 10-dimension LPC features can achieve very similar results as that gained from SVM without feature selection. By using 30-second multi-clip classification and post-processing on classification result, the classification accuracy is improved from 47.4% to 75.2%, which is higher than that professional people got on music clip.

关键词： Support vector machines Support vector machine classification Music information retrieval Sampling methods Rhythm Mood Computer science Feature extraction Mel frequency cepstral coefficient linear predictive coding

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Instrument Identification in Monophonic Music Using Spectral Information

Instrument Identification in Monophonic Music Using Spectral...

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

作者： Mizuki Ihara Shin-ichi Maeda Shin Ishii Graduate School of Information Science Graduate School of Informatics Nara Institute of Science and Technology Ikoma Nara Japan Graduate School of Informatics Kyoto University Kyoto Japan

Various kinds of feature sets have been proposed to represent characteristics of musical instruments. While those feature sets have been chosen in a rather heuristic way, in this study, we demonstrate that the log-power spectrum suffices to represent characteristics that are essential to identifying instruments. For efficient encoding of instrument characteristics, we then reduce the number of features by applying the well-known dimension reduction techniques: principal component analysis (PCA) and linear discriminant analysis (LDA). For the classification of eight instruments, the features obtained by applying PCA-LDA to the log-power spectrum performed very well in comparison to existing methods with a recognition rate of 91% with as few as ten features.

关键词： Instruments linear discriminant analysis Cepstral analysis Principal component analysis Feature extraction Signal processing linear predictive coding Music Frequency Loudspeakers

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Closed Loop Dynamic Bit Allocation for Excitation Parameters in Analysis-by-Synthesis Speech Codec

Closed Loop Dynamic Bit Allocation for Excitation Parameters...

引用

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

作者： James P. Ashley Udar Mittal Speech Processing Research Laboratory Motorola Research Laboratories Schaumburg IL USA

A method for dynamically allocating bits for the adaptive and fixed codebook of an analysis-by-synthesis speech codec is proposed. The bit allocation is based on the closed loop weighted mean squared error. The different bit allocations identify various codebook configurations used by the adaptive and fixed codebooks of the codec. Unlike open loop approaches where the decision on a codebook configuration is made once per frame, in the closed loop approach the decision on which codebook configuration should be used is made in each subframe. A variable length code is used for coding the codebook configuration. The factorial packing codebook is used as the fixed codebook. The technique is a part of 8.5 kbps mode of EVRC-WB speech coding standard.

关键词： Bit rate Speech analysis Speech codecs Speech synthesis Delay Nonlinear filters Speech coding Adaptive filters linear predictive coding Speech processing

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A Fixed Dimension Modified Sinusoid Model (FD-MSM) for Single Microphone Sound Separation

A Fixed Dimension Modified Sinusoid Model (FD-MSM) for Singl...

引用

IEEE International Conference on Signal Processing and Communications, ICSPC

作者： Pejman Mowlaee Begzade Mahale Abolghasem Sayadiyan Alireza Bayesteh Tashk Department of Electrical Engineering Amirkabir University of Technology슠 Tehran Iran

The lack of a flexible analysis model has been introduced as an important issue in different applications like source separation. In this paper, a fixed dimension modified sinusoid model (FD-MSM) is proposed for analysis of all audible signals consisting of speech, music and their mixtures. Employing the peak picking in Meldomain gives rise to a fixed number of parameters in the proposed FDMSM, which is desired in clustering algorithms like VQ (vector quantization) or GMM (gaussian mixture model), commonly used for source separation scenarios. Applying the proposed FD-MSM to various audible signals, it is observed that the resulting signal is perceptually indistinguishable from the original.

关键词： Microphones Speech analysis Signal analysis Source separation Speech synthesis Multiple signal classification Vector quantization linear predictive coding Power harmonic filters Computational complexity

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