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检索条件"主题词=Linear Predictive coding"

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A comparison of the new 2400 bps MELP Federal Standard with other standard coders

A comparison of the new 2400 bps MELP Federal Standard with ...

引用

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

作者： M.A. Kohler U.S. Department of Defense Fort Meade MD USA

In 1996, the U.S. Department of Defense Digital Voice Processing Consortium (DDVPC) selected Texas Instrument's mixed excitation linear prediction (MELP) algorithm as the recommended new Federal Standard for 2400 bps voice communications. The algorithm selection process involved quality, intelligibility, communicability, and recognizability testing in many acoustic noise, error, and tandem conditions. Algorithm complexity was also measured. This paper compares the performance scores, diagnostic information, and complexity of MELP to the 4800 bps Federal Standard (FS1016) code excited linear prediction (CELP) algorithm, the 16 kbps continuously variable slope delta modulation (CVSD) algorithm, and the venerable Federal Standard (FIPS Pub. 137) 2400 bps linear predictive coding (LPC-10) algorithm.

关键词： Code standards Acoustic noise Prediction algorithms Acoustic testing linear predictive coding Delta modulation Working environment noise Microphones Instruments Communication standards

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On scalability in CELP coding systems

On scalability in CELP coding systems

引用

IEEE Workshop on Speech coding For Telecommunications

作者： R. Taori R.J. Sluijter A.J. Gerrits Signal Processing Group Philips Research Laboratories Eindhoven Netherlands

We propose methods to achieve bit-rate and complexity scalability in CELP coders/decoders. These methods can be implemented as plug-ins or add-ons to existing CELP systems. The associated scale factors can be readjusted even during run-time. In particular, using the proposed methods, a wideband (sampling rate 16 kHz) coder/decoder has been developed that is capable of reconstructing good to high quality speech at bit rates ranging from 14 to 24 kbit/s. The possible complexity range of the decoder can amount to a factor of 2 or more, depending on the constitution of the transmitted bitstream. Although the system also incorporates a complexity scalable encoder, it is not discussed in this paper. This system has been submitted to MPEG-4, where it has been incorporated in the MPEG-4 Audio Verification Model.

关键词： Scalability Decoding linear predictive coding Interpolation Bit rate Reflection Computational complexity Wideband Speech synthesis Degradation

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High quality split band LPC vocoder operating at low bit rates

High quality split band LPC vocoder operating at low bit rat...

引用

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

作者： I. Atkinson S. Yeldner A. Kondoz Centre for Communication Systems Research University of Surrey Guildford Surrey UK University of Surrey Guildford Surrey GB

LPC based speech coders operating at bit rates below 3.0 kbits/sec are usually associated with buzzy or metallic artefacts in the synthetic speech. These are mainly attributable to the simplifying assumptions made about the excitation source, which are usually required to maintain such low bit rates. A new LPC vocoder is presented which splits the LPC excitation into two frequency bands using a variable cut-off frequency. The lower band is responsible for representing the voiced parts of speech, whilst the upper band represents unvoiced speech. In doing so the coder's performance during both mixed voicing speech and speech containing acoustic noise is greatly improved, producing soft natural sounding speech. The paper also describes new parameter determination and quantisation techniques vital to the operation of this coder at such low bit rates.

关键词： linear predictive coding Vocoders Bit rate Speech enhancement Equations Cutoff frequency Acoustic noise Quantization Gaussian noise Robots

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Implementation of a real-time HY-2 channel vocoder algorithm

Implementation of a real-time HY-2 channel vocoder algorithm

引用

MILCOM, Military Communications Conference

作者： T.S. Loos Raytheon Electronics Systems Richardson TX USA

This paper describes the proprietary implementation of a real-time software based HY-2 compatible channel vocoder algorithm. The HY-2 algorithm was used prior to linear predictive coding (LPC) vocoders for low bit rate voice communications at 2400 bps. This implementation is a unique and instructive implementation of a parametric vocoder, producing high quality speech.

关键词： Vocoders Filter bank Synthesizers linear predictive coding Channel bank filters Software algorithms Digital signal processing Speech processing Bit rate Decoding

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A hierarchical bounding box method for searching the speech database for speech recognition

A hierarchical bounding box method for searching the speech ...

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IEEE International Symposium on Consumer Electronics (ISCE)

作者： Wai Tong Fung Shiu Yin Yuen Chi Hang Tse Department of Electronic Engineering City University of Hong Kong Hong Kong China

ISBN: (纸本)0780343719

Speech recognition is an increasingly popular method for Chinese character input. A fast and reliable hierarchical bounding box method for searching the speech database is proposed. The method borrows from ideas in computer graphics, where the hierarchical bounding box concept is used for fast ray-object intersection tests (Hearn et al. 1997).

关键词： Databases Speech recognition linear predictive coding Computer graphics Testing Robustness Writing Data engineering Nearest neighbor searches predictive models

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Efficient quantization of LSF by utilizing dynamic interpolation

Efficient quantization of LSF by utilizing dynamic interpola...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Q. Zhao J. Suzuki Department of Information and Computer Science Saitama University Urawa Saitama Japan

Utilizing the ordering property of LSF (Line Spectral Frequency) parameters, we propose a new quantization method for scalar quantization of LSF coefficients, which uses a new set of quantization parameters called LSFI instead of direct LSF or Differential LSF (LSFD). Based on the experimental analysis the new quantization method is further improved by combining with LSFD parameters (we call this improved version an LSFID scheme). The experimental results have shown that the proposed methods are effective in decreasing quantization distortion. Compared with LSFD, the LSFID quantizer can save at least 1 bit per frame to achieve 1dB quantization.

关键词： Quantization Interpolation Frequency Speech coding Bit rate linear predictive coding Equations Computer simulation Telephony Reflection

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On a spectral scaling technique for pitch control in the high-quality text-to-speech type system

On a spectral scaling technique for pitch control in the hig...

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： JuimgGpie Cjimg MyungJin Bae Dept. of Telecommun. Eng. Soongsil Univ. Seoul South Korea Dpt. of Telecommunication Engr Soongsil University Seoul South Korea

ISBN: (纸本)0780336941

Speech synthesis technique is classified into three groups; waveform coding, source coding and hybrid coding. Waveform coding and hybrid coding are used for sentence-based-synthesis by synthesis-by-analysis method; the difficulty of pitch altering makes them inappropriate for synthesis-by-rule. However, if it is possible to alter the pitch period when the waveform coding is used, synthesis-by-rule is available maintaining good intelligibility and naturalness comparable to the original speech. In this paper, we propose a new pitch alteration method that can change the pitch period in waveform coding by scaling the time-axis and where phase compensation is performed by using the zero-inserting and pitch-halving method.

关键词： Speech synthesis Speech coding Frequency domain analysis Time frequency analysis Cepstrum Telecommunication control Source coding Very large scale integration Packaging linear predictive coding

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CD-HMM algorithm performance for speaker identification on an Italian database

CD-HMM algorithm performance for speaker identification on a...

引用

International Conference on Information, Communications and Signal Processing

作者： C. Caini P. Salmi A.V. Coralli Dipartimento di Elettronica Informatica e Sistemistica University of Bologna Bologna Italy

The focus of this work is on the performance analysis of a text dependent closed set speaker identification system for the Italian language. Two identification algorithms, based on LPC and LPC-cepstral feature extractors followed by a continuous density hidden Markov model (CD-HMM) classifier, have been implemented and tested on the Italian database SIVA the MUSER. The database consists of 360 phone calls made by 20 different male speakers from different Italian regions. The false identification probability for the two algorithms has been evaluated for different training sets, different spoken words and a variable number of states of the CD-HMM classifier. Results show that, in any of the considered conditions, the LPC-cepstral based system performs better than the LPC based one and that, in the best working condition, the false identification probability turns out to be of the order of 1.5 per cent.

关键词： Speech analysis linear predictive coding Spatial databases Feature extraction Hidden Markov models Employee welfare Speaker recognition System testing Performance analysis Natural languages

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Study of the near Si-SiO/sub 2/ interface trap layer using the charge pumping technique

Study of the near Si-SiO/sub 2/ interface trap layer using t...

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International Conference on Semiconductor

作者： Y. Maneglia D. Bauza UMR CNRS 5531 LPCS ENSERG/INPG Grenoble France

It is shown that the charge pumping technique allows the extraction of the Si-SiO/sub 2/ interface depth trap concentration profile. This profile is found of the form N/sub t/(x)=N/sub ts/ exp(-x/d)+N/sub to/ where d is the distance in the oxide from the interface. The method is discussed as well as the expression of the trap density obtained.

关键词： Charge pumps Filling Frequency Equations linear predictive coding MOS devices Current measurement Charge measurement MOSFETs Density measurement

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A review of algorithms for perceptual coding of digital audio signals

A review of algorithms for perceptual coding of digital audi...

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International Conference on Digital Signal Processing (DSP)

作者： T. Painter A. Spanias Department of Electrical Engineering Telecommunications Research Center Arizona State University Tempe AZ USA

Considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. First, psychoacoustic principles are described with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Then, we review methodologies which achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms which manipulate transform components and subband signal decompositions. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms which have become international and/or commercial standards are also presented, including the ISO/MPEG family and the Dolby AC-3 algorithms. The paper concludes with a brief discussion of future research directions.

关键词： Audio coding Costs Psychology Psychoacoustic models ISO standards Pulse modulation Multimedia systems linear predictive coding DVD Multimedia computing

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