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检索条件"主题词=Linear predictive coding"

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On the degradation features of poly-emitter n-p-n BJTs after hot carrier injection

On the degradation features of poly-emitter n-p-n BJTs after...

引用

Annual International Symposium on Reliability Physics

作者： F. Maugain C. Papadas G. Ghibaudo N. Gambetta P. Mortini Central Research and Development SGS-THOMSON Microelectronics Italy Central Research and Development SGS-THOMSON Microelectronics France Inst of Microelectronics National Centre for Scientific Research Attica Greece LPCS-ENSERG Grenoble France

The purpose of this paper is to comment on the degradation features of poly-emitter n-p-n BJTs after hot carrier injection. First, a physical model which quantitatively explains the experimentally observed gain degradation is presented and its suitability for Computer Aided Reliability (CAR) applications is demonstrated. Besides, a "charge-pumping" based technique, allowing the accurate evaluation of the total fast electronic Si/SiO/sub 2/ interface states located at the perimeter of the emitter-base junction is also proposed.

关键词： Degradation Hot carrier injection Human computer interaction Interface states Microelectronics BiCMOS integrated circuits Stress Tunneling Research and development linear predictive coding

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Pattern playback from 1950 to 1995

Pattern playback from 1950 to 1995

引用

IEEE International Conference on Systems, Man and Cybernetics

作者： M. Slaney Interval Research Corporation Palo Alto CA USA

This paper describes algorithms to convert spectrograms, cochleagrams and correlograms back into sounds. Each of these representations converts sound waves into pictures or movies. Techniques for inversion, known as the pattern playback problem, are important because they allow these representations to be used for analysis and transformations of sound. The algorithms described here use convex projections and intelligent phases guesses to iteratively find the closest waveform consistent with the known information. Reconstructions from the spectrogram and cochleagram are indistinguishable from the original sound. In informal listening tests, the correlogram reconstructions are nearly identical.

关键词： Spectrogram Motion pictures Psychoacoustic models Image reconstruction Testing Milling machines Pattern analysis Iterative algorithms Humans linear predictive coding

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A full-custom extensible and compact digital multiplier circuit design

A full-custom extensible and compact digital multiplier circ...

引用

International Conference on Semiconductor

作者： A.C. Negoi Electronics and Telecommunications Department Politehnica University of Bucharest Grenoble France

This paper presents a design of an extensible and very compact digital multiplier, made in full-custom. It is characterized by regularity and it is particularly adapted to be used as a part of high-speed digital proce... 详细信息

关键词： Circuit synthesis Circuit simulation Equations linear predictive coding Process design Arithmetic Silicon Delay Inverters Logic circuits

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A Comparison of entropy coding performances between scalar and vector quantized speech LPC parameters

A Comparison of entropy coding performances between scalar a...

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IEEE Workshop on Speech coding For Telecommunications

作者： Junchen Du Polytechnic University

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Efficient DSP design for vocoder application

Efficient DSP design for vocoder application

引用

Custom Integrated Circuits Conference (CICC)

作者： H.Y. Yoo J.J. Kim K.J. Byun K.C. Han D.K. Kim J.S. Kim H.B. Lee M.J. Bae Semiconductor Division Electronics and Telecommunications Research Institute Taejon South Korea Electronics and Telecommunications Research Institute Daejeon Daejeon KR Semicond. Div. ETRI Taejon South Korea

Vocoder is a key component of the digital cellular system. A DSP architecture which supports the direct and immediate addressing modes in one instruction cycle, combined with a RISC-style instruction set, turns out to be effective for vocoder implementation in terms of proper performance and lower power consumption requirements. By adopting a dual bank memory system and an efficient ALU bus scheme, two 16-bit operand access and ALU operation can be executed simultaneously. Improved implementation methods applicable to CELP type vocoder in which the pitch search is performed by the analysis-by-synthesis, are also presented.

关键词： Digital signal processing Vocoders Energy consumption Delay effects Multiaccess communication linear predictive coding Algorithm design and analysis Encoding Decoding Data mining

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Speaker modification with LPC pole analysis

Speaker modification with LPC pole analysis

引用

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

作者： J. Slifka T.R. Anderson Systems Research Laboratories Inc. Dayton OH USA Armstrong Laboratory AL/CFBA OH USA

Speaker modification is the ability to change the perceived speaker identity of a recorded utterance. Basic to this is the capability to alter the vowel segments of speech. Not only do these segments comprise the majority of the voiced portion of speech but they are dominated by clearly defined acoustic parameters-formant frequencies and pitch. A method of altering the formant frequencies of vowel segments using LPC analysis/synthesis was investigated. Pole location modification based on statistical references provided individual control over formant frequencies and bandwidths but, in some transformations, lead to artifacts in the reconstructed speech.

关键词： linear predictive coding Loudspeakers Speech synthesis Psychoacoustic models Frequency Filters Laboratories Polynomials Resonance Performance analysis

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Efficient coding of LSP parameters using split matrix quantisation

Efficient coding of LSP parameters using split matrix quanti...

引用

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

作者： C.S. Xydeas C. Papanastasiou Speech Processing Research Laboratory Electrical Engineering Division School of Engineering University of Manchester Institute of Science and Technology Manchester UK

This paper presents a new and efficient LPC quantisation scheme called split matrix quantisation (SMQ). The proposed method can be viewed as an extension of the conventional split vector quantisation process. It operates over N consecutive LPC frames and effectively divides a p/spl times/N LSP matrix into K submatrices which are then vector quantised independently. SMQ exploits the interframe redundancy that exists between consecutive sets of LSP coefficients and achieves "transparent" quantisation at 900 bits/sec. "High quality" LSP quantisation can be easily obtained at 750 bits/sec. These bit rates are based in a 20 msec LPC analysis frame size. Furthermore, SMQ is characterised by relatively low complexity and low storage requirements.

关键词： linear predictive coding Bit rate Filters Books Distortion measurement Speech processing Vector quantization Delay Speech synthesis Laboratories

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A cepstrum chip: architecture and implementation

A cepstrum chip: architecture and implementation

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： An-Nan Suen Jhing-Fa Wang Yuen-Lin Chiang Institute of Information Engineering National Cheng Kung University Tainan Taiwan

The cepstrum coefficients have been widely used for speech signal representation and play a very important role in recognition accuracies. We present a low cost architecture for VLSI implementation of LPC-based cepstrum algorithm. The circuit performs the cepstrum operation for each frame of the speech data. A pipelining architecture leads to high speed performance up to speech recognition rate. The cepstrum chip is fabricated in 1.2 /spl mu/m double-metal CMOS technology after the physical design and circuit verification. On the whole, the chip can process 18.3 MHz sampled data and it contains about 24000 transistors which occupy 227.5/spl times/213.3 mils/sup 2/ area. It has been shown to be fully functional and is the first working cepstrum chip.

关键词： Cepstrum Switches Speech recognition Circuit testing linear predictive coding CMOS technology Fourier transforms Cepstral analysis Hidden Markov models Flow graphs

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A continuous speech recognition system using finite state network and Viterbi beam search for the automatic interpretation

A continuous speech recognition system using finite state ne...

引用

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

作者： Nam-Yong Han Hoi-Rin Kim Kyu-Woong Hwang Young-Mok Ahn Joon-Hyung Ryoo Automatic Interpretation Section Electronics and Telecommunications Research Institute USA

ISBN: (纸本)0780324315

This paper describes a Korean continuous speech recognition system using phone based semi-continuous hidden Markov model (SCHMM) method for automatic interpretation. The task domain is hotel reservation. The system (composed of speech recognition, machine translation and speech synthesis) has the following three features. First, an embedded bootstrapping training method is used that enables us to train each phone model without the need for a phoneme segmentation database. Second, a hybrid estimation method which is composed of the forward-backward algorithm and the Viterbi algorithm is proposed for the HMM parameter estimation. Third, a between-word modeling technique is used at the function word boundaries. The recognition results in speaker independent experiments are as follows. In the case of Version 1, the continuous speech recognition result is 89.1% and in Version 2, the result is 97.6%.

关键词： Speech recognition Viterbi algorithm Gold Hidden Markov models Vocabulary linear predictive coding Dictionaries Cepstral analysis Electronic mail Spatial databases

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High-quality audio-coding at less than 64 kbit/s by using transform-domain weighted interleave vector quantization (TwinVQ)

High-quality audio-coding at less than 64 kbit/s by using tr...

引用

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

作者： N. Iwakami T. Moriya S. Miki Human Interface Laboratories NTT Musashino Tokyo Japan

A new audio-coding method is proposed. This method is called transform-domain weighted interleave vector quantization (TwinVQ) and achieves high-quality reproduction at less than 64 kbit/s. The method is a transform coding using modified discrete cosine transform (MDCT). There are three novel techniques in this method: flattening of the MDCT coefficients by the spectrum of linear predictive coding (LPC) coefficients; interframe backward prediction for flattening the MDCT coefficients; and weighted interleave vector quantization. Subjective evaluation tests showed that the quality of the reproduction of TwinVQ exceeded that of an MPEG Layer II coder at the same bitrate.

关键词： linear predictive coding Vector quantization Discrete cosine transforms Transform coding Bit rate Power harmonic filters Frequency domain analysis Testing Computational complexity Signal design

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