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2nd International Conference on the Applications of Digital Information and Web Technologies

作者： Merouane, Bouzid USTHB Elect Fac Speech Commun & Signal Proc Lab Algiers 16111 Algeria

ISBN: (纸本)9781424444564

In this paper, an optimized trellis coded vector quantization (OTCVQ) system designed for efficient and robust coding of LSF spectral parameters is presented. The aim of this system, called at the beginning "LSF-OTCVQ Encoder", is to achieve a low bit rate transparent quantization of the FS1016 LSF parameters. Once the effectiveness of the LSF-OTCVQ encoder was proven in the case of ideal transmissions over noiseless channel, we were interested after in the improvement of its robustness for real transmissions over noisy channel. To protect implicitly the transmission indices of the LSF-OTCVQ encoder incorporated in the FS1016, we used a joint source-channel coding carried out by the channel optimized vector quantization.

关键词： Robustness Quantization linear predictive coding Oral communication Decision support systems Fiber reinforced plastics Virtual reality

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ADAPTIVE POSTFILTER IN 16KBPS LD-CELP SPEECH CODER

ADAPTIVE POSTFILTER IN 16KBPS LD-CELP SPEECH CODER

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1996 3rd International Conference on Signal Processing(ICSP’96)

作者： Wang Bingxi He Yinghua Zhengzhou Henan China

ISBN: (纸本)0780329120

＜正＞In September 1992,the recommendation G 728,which is a 16kbps LD-CELP speech coder submitted by AT＆Twas standarized by *** the process of ratification test[1],the coder’s performances were equivalent to or better than that of 32kbps ADPCM for all conditions *** paper,which is based on a G.728 encoding-decoding system simulated in software,studies and tests different parts of the algorithm,espacially that of the postfitter

关键词： adaptive filters speech coding linear predictive coding filtering theory adaptive postfilter LD-CELP speech coder G.728 encoding-decoding system CCITT 16 kbit/s

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Candidate scheme for MP-MLQ search in G.723.1 3

Candidate scheme for MP-MLQ search in G.723.1

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3rd IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC 01)

作者： Chen, FK Yang, JF Natl Cheng Kung Univ Dept Elect Engn Tainan 70101 Taiwan

ISBN: (纸本)0780367200

For the purpose of low-cost or low power consumption, a cost-effective solution with good trade-off between quality and complexity must be estimated for small mobile machine. In this paper, we propose a condensed MP-MLQ algorithm to reduce search computation required for G723.1. Simulation results show that the proposed method can reduce about 50 similar to 82% computational load for stochastic codebook search with perceptually intangible degradation. This reductionism would be helpful to save the power consumption and relax CPU load for the mobile-machine in wireless communications, e.g. PALM, PDA.

关键词： Computational modeling Degradation Distributed computing Energy consumption GSM linear predictive coding Quantization Speech analysis Speech coding Stochastic processes

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Navigation Autopilot With Real Time Voice Command Recognition System

Navigation Autopilot With Real Time Voice Command Recognitio...

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21st Signal Processing and Communications Applications Conference (SIU)

作者： Tezer, Huseyin Kursat Yagimli, Mustafa Deniz Harp Okulu Elekt & Elekt Muhendisligi Bolumu Istanbul Turkey

The navigation autopilot system was developed using real-time voice command recognition system for the ship open water cruise. In this design, linear predictive coding (LPC) algorithm is used for the extraction of the... 详细信息

ISBN: (纸本)9781467355636;9781467355629

关键词： navigation autopilot voice command recognition linear predictive coding dinamic time warping

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Development of Concatenative Syllable-Based Text to Speech Synthesis System for Tamil

Development of Concatenative Syllable-Based Text to Speech S...

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International Conference on Artificial Intelligence and Evolutionary Algorithms in Engineering Systems (ICAEES)

作者： Sudhakar, B. Bensraj, R. Annamalai Univ Dept Elect Engn Annamalainagar Chidambaram India

ISBN: (纸本)9788132221265;9788132221258

This paper addresses the problem of improving the intelligibility of the synthesized speech in Tamil text-to-speech (TTS) synthesis system. The human speech is artificially generated by speech synthesis. The normal language text will be automatically converted into speech using TTS system. This paper deals with a corpus-driven Tamil TTS system based on the concatenative synthesis approach. Concatenative speech synthesis involves the concatenation of the basic units to synthesize an intelligent, natural sounding speech. In this paper, syllables are the basic unit of speech synthesis database and the modification of syllable pitch by timescale modification. The speech units are annotated with associated prosodic information about each unit, manually or automatically, based on an algorithm. An annotated speech corpus utilizes the clustering technique that provides way to select the suitable unit for concatenation, depending on the minimum total joint cost of the speech unit. The entered text file is analyzed first, this syllabication is performed based on the linguistics rules, and the syllables are stored separately. Then, the syllable corresponding speech file is concatenated and the silence present in the concatenated speech is removed. After that, discontinuities are minimized at syllable boundaries without degrading the quality. Smoothing at the concatenated syllable boundary is performed, changing the syllable pitches by timescale modification.

关键词： linear predictive coding TTS synthesis system Corpus-driven Tamil TTS system PCM

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Temporal Compression in Wireless Sensor Networks using Compressive Sensing and ARMA modeling 2

Temporal Compression in Wireless Sensor Networks using Compr...

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2nd International Conference on Advances in Computing, Communication, and Automation (ICACCA) (Fall)

作者： Thapliyal, Ashish Kumar, Rajender Natl Inst Technol Kurukshetra Dept Elect & Commun Engn Kurukshetra Haryana India

ISBN: (纸本)9781509034802

In this paper a new method is proposed which reduce the redundancy of data transmission in Wireless Sensor Network (WSN). In a dense environment of sensors, energy efficiency is key factor for prolonged battery life. Here in this paper temporal redundancy of sensor node is reduced using hybrid of two methods, one is compressive sensing using Discrete Cosine Transform (DCT) and other one is linear predictive coding (LPC). Given method is not suitable for real time data monitoring rather it is better for delay tolerant services as it require processing time before transmission. Here it is assumed that data monitored by sensors is piecewise linear. The relevancy in temporal data exploited here for data compression. First data monitored from each node for a long enough span of time which is further segmented according to their linearity. As compressive sensing compress the data and ARMA also use only few of the transmission bits, combination of both provide satisfactory compression of stored data. The segmented data used as input for compression and further processed to sink node. Result shows that this scheme provide satisfactory compression rate of monitored data.

关键词： wireless sensor network Discrete cosine transform linear predictive coding Compressive Sensing, auto regressive moving average

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Acoustical Detection of Gunshots 25

Acoustical Detection of Gunshots

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25th International Conference RADIOELEKTRONIKA

作者： Hrabina, Martin Sigmund, Milan Brno Univ Technol Dept Radio Elect Brno Czech Republic

ISBN: (纸本)9781479981175

This paper describes development of reliable gunshot detection system with emphasis on low power consumption for use in counter-poacher devices primarily protecting elephants in Africa. Intended system will work as a binary detector of gunfire without further classification of used fire-arm. Dominance of right gunshot detection over false alarms is crucial. Proposed recognition system is based on linear predictive coefficients, correlation against template and comparison of spectral energy in sub-bands.

关键词： gunshot recognition signal processing linear predictive coding

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Pre-processing of noisy speech for voice coders

Pre-processing of noisy speech for voice coders

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IEEE Speech coding Workshop

作者： Agarwal, T Kabal, P McGill Univ Dept Elect & Comp Engn Montreal PQ H3A 2A7 Canada

ISBN: (纸本)0780375491

Accurate linear Prediction Coefficient (LPC) estimation is one of the key requirements for low bit-rate voice coding. Under harsh acoustic conditions, LPC estimation can become unreliable. This results in poor quality of encoded speech and introduces annoying artifacts. This paper presents a two-branch speech enhancement preprocessing scheme for low bit-rate voice coders. This scheme consists of two parallel denoising blocks. One block will enhance the degraded speech for LPC estimation. Another block will increase the perceptual quality of the speech to be coded. The goal of this paper is to design the two-branch scheme. Test results show that the two-branch scheme can provide better perceptual quality compared to conventional one-branch speech enhancement techniques in noisy environments.

关键词： speech enhancement parallel two-branch speech enhancement linear prediction coefficient estimation voice coders noisy speech preprocessing signal denoising parameter estimation vocoders speech coding random noise linear predictive coding Vocoders Speech enhancement signal denoising speech coding random noise Parameter estimation Parallel Lines Speech

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PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE HEARING-IMPAIRED SPEAKERS

PHONETIC AND INTONATION ENHANCEMENT OF SPEECH FOR JAPANESE H...

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IEEE International Conference on Multimedia and Expo (ICME 2008)

作者： Watanabe, Shogo Moriyama, Tsuyoshi Ozawa, Shinji Keio Univ Dept Informat & Comp Sci Yokohama Kanagawa 2238522 Japan

ISBN: (纸本)9781424425709

We propose a system that is capable of improving intelligibility of speech by those with hearing impairment. We have found the speech often had problems in the intonation, the duration of the unvoiced consonants, and the tone of the voiced phonemes. The system systematically compensates problematic components of the speech using the counterparts in normal speech. It corrects the intonation using TD-PSOLA and elongates consonants by repeating the original waveform using inverting technique for making the result continuously connected until the duration reaches a threshold. Experimental results show that the proposed method successfully improves the intelligibility of the speech from 28% to 35% by making phonemes more articulated, and making double consonants, V of syllabic consonant, and unvoiced consonants perceived clearer.

关键词： Speech analysis Speech intelligibility Speech enhancement Speech synthesis linear predictive coding

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The Development of “Solid State Speech” Technology at Texas Instruments

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IEEE Acoustics, Speech, and Signal Processing Newsletter 1981年第1期53卷 34-38页

作者： Frantz, Gene Wiggins, Richard Texas Instruments United States

The first single chip speech synthesizer was introduced by Texas Instruments in June of 1978, and forms the basis for the very popular “Speak & Spell”* learning aid. The development of this unique integrated circuit was due in a large measure to a well-defined product's need for a specific new semiconductor capability. A cost effective solution called for a novel circuit design to cope with the complex digital signal processing required for speech synthesis. The resulting technology breakthrough has resulted in the introduction of a number of voice IC's and the rapid development of voice synthesis markets. Among the important lessons of the Speak & Spell* product, two stand out. The first is the potential impact of custom designed VLSI circuitry. The speech synthesis algorithm implemented in the Speak & Spell* speech chip is but one of many useful processing tasks that are considered ″too complex″ for general purpose processors. Semiconductors with special purpose processing architectures will surely bring many other exciting 0IC capabilities. The second lesson concerns the assessment of the impact of new technology. The task of predicting market acceptance for a product with revolutionary technology offering truly new functions, tested present marketing approaches. It wasn't always clear that the product would be successful or that its new Solid State Speech* technology would be perceived as useful. © 1981, IEEE. All rights reserved.

关键词： Solid state circuits Speech synthesis linear predictive coding Synthesizers Consumer products Speech processing Digital filters

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