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检索条件"主题词=linear predictive coding"

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An embedded-code multirate speech transform coder

An embedded-code multirate speech transform coder

引用

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

作者： M. Berouti J. Makhoul Bolt Beranek and Newman 슠Inc. Cambridge MA USA

In embedded multirate speech coding, it is desired to have a transmitter/receiver system that operates efficiently over a wide range of channel transmission rates. We are currently investigating a system based on adaptive transform coding of speech, in which we code and transmit the system parameters and the discrete cosine transform (DCT) coefficients of the fullband linear prediction residual waveform. The multirate property of the system is achieved by allowing the channel to discard some of the bits generated by the high data rate transmitter. Stripping off bits results in an absence of DCT components, which the receiver regenerates by a spectral duplication method. An inverse DCT at the receiver yields the time domain residual waveform to be used as input to the linear prediction synthesis filter. The lowest data rate achievable by the system is about 2.5 kb/s, in which case the system reduces to a narrowband LPC pitch-excited vocoder.

关键词： Discrete cosine transforms Transmitters Speech coding Filters Speech synthesis linear predictive coding Bit rate Frequency Transform coding Narrowband

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Word boundary detection by pitch contours in an artificial language

Word boundary detection by pitch contours in an artificial l...

引用

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

作者： Y. Kobayashi Y. Niimi Department of Computer Science Kyoto Technical University Kyoto Japan

The pitch contour plays an important role of marking syntactic boundaries in the natural language. In order to improve the continuous speech recognition system we have developed, we have carried out experiments on how the pitch contour reflects word boundaries in an artificial language-continuously read material of a programming language. On the analysis of 173 sentences of 5 male speakers including 1655 word boundaries, 67.1 % of the total word boundaries were correctly detected and the false alarm is 28.8 %. The preceding and the following boundaries of keywords were found at the correct rate of 94.2 % and 85.0 % respectively. The application of boundary information to the recognition of continuous speech is also discussed in this paper.

关键词： Speech recognition Cepstrum Computer science Natural languages Computer languages Stress Loudspeakers Filter bank linear predictive coding Explosions

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Effect of corruption within the recursive estimation of spectral parameters for LPC

Effect of corruption within the recursive estimation of spec...

引用

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

作者： R. Wiggins J. Parry Central Research Laboratories Texas Instruments Inc. Dallas TX USA Department of Electrical Engineering University of Arizona Tucson Tucson AZ USA

Inverse filter parameters may be determined from estimates of the autocorrelation or covariance of a sampled signal by methods which admit corruption of partial results and generate subsequently optimal values for the remaining parameters. Experimental results are presented to indicate the extent to which such techniques can compensate for the effects of irregular or very gross parameter quantization. Three-way comparisons are made between the spectral responses of the inverse filters designed with the globally optimal parameters, corrupted versions of those parameters and parameters determined with the corruption introduced within the estimation algorithm.

关键词： Recursive estimation linear predictive coding Equations Reflection Autocorrelation Lattices Nonlinear filters Quantization Polynomials

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A maximum peakiness criterion for deconvolving speech waveforms

A maximum peakiness criterion for deconvolving speech wavefo...

引用

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

作者： S. Maitra S. Foster C. Davis Systems Control Technology Inc. Palo Alto CA USA

linear predictive coding of speech has traditionally used a least mean square (LMS) criterion for determining the optimal filter parameters. Auto-regressive (AR) models using LMS are in wide-spread use for low bit rate speech compression. These models are particularly susceptible to moderate amounts of added noise and distortion. This paper proposes a new model which does not use the least-mean-square criterion but rather uses the "peakiness" of the deconvolved waveform to determine the filter parameters. Since voiced output speech is reproduced with a maximally "peaked" waveform, viz, an impulse, this criterion is naturally suited for the characteristics of the synthesis process. Preliminary analysis also shows the feasability of implementing the proposed algorithm in microprocessor based hardware.

关键词： Least squares approximation Speech synthesis linear predictive coding Speech coding Nonlinear filters Bit rate Speech processing Algorithm design and analysis Microprocessors Hardware

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The enhancement of wordspotting techniques

The enhancement of wordspotting techniques

引用

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

作者： R. Wohlford A. Smith M. Sambur ITT Defense Communications Division San Diego CA Defense Communications Division ITT San Diego CA USA

This paper presents results of research on the enhancement of speaker independent wordspotting in conversational, telephone bandwidth speech from a variety of talkers. The research involved the comparison of five LPC based parameter sets (filter coefficients, autocorrelation coefficients, cepstral coefficients, vocal tract area functions, and pseudoformants), in three levels of additive white noise, and the evaluation of two methods of noise reduction. The techniques were studied with a dynamic programming based wordspotting system. The performance of the system was measured the percent of keywords spotted with no false alarms. The results show that pseudoformants, without the use of speaker normalization or noise reduction techniques, perform as well as or better than the other parameter sets with these techniques.

关键词： Noise reduction Telephony Bandwidth Speech enhancement linear predictive coding Filters Autocorrelation Cepstral analysis Additive white noise Dynamic programming

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The implementation of an all digital speech synthesizer using a multimicroprocessor architecture

The implementation of an all digital speech synthesizer usin...

引用

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

作者： C. Hodges T. Barnwell D. McWhorter School of Electrical Engineering Georgia Institute of Technology Atlanta GA USA

This paper presents a unique approach for the implementation of the speech synthesis portion of a LPC vocoder. The implementation uses eight LSI-11 microprocessors operating synchronously around a time division multiplexed multiport memory. The implementation of the recursive digital filter portion of the LPC synthesis is accomplished without the use of any high speed external arithmetic elements and in a way which requires no synchronization overhead. This particular technique illustrates a general procedure which is applicable to the implementation of a large class of digital signal processing algorithms on multiprocessor machines.

关键词： Speech synthesis Synthesizers linear predictive coding Vocoders Microprocessors Time division multiplexing Digital filters Signal synthesis Digital arithmetic Digital signal processing

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A scheme for concatenating units for speech synthesis

A scheme for concatenating units for speech synthesis

引用

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

作者： J. Olive Bell Telephone Laboratories Inc. USA

In previous papers we discussed a speech synthesis by rule scheme where segments obtained from natural speech were linearly concatenated. These segments included the consonants and the transitions from consonants to vowels, vowels to vowels, and vowels to consonants. Each synthesis parameter was defined by few sets of LPC area parameters, and in the concatenative process, straight line interpolation was used to obtain the complete set of area parameters. Informal listening and some formal intelligibility testing revealed that this simplified description of the synthesis segments was not sufficient to produce the speech quality that would satisfy us. Consequently, it was decided to improve the definition of the concatenative units. This paper will discuss in detail the scheme for synthesis employing these concatenative units.

关键词： Speech synthesis linear predictive coding Natural languages Testing Steady-state Telephony Laboratories Concatenated codes History Smoothing methods

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Spectral distance measure applied to the optimum design of DPCM coders with L predictors

Spectral distance measure applied to the optimum design of D...

引用

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

作者： J.-P. Adoul J.-L. Debray D. Dalle Communication Research Center University of Sherbrooke Sherbrooke QUE Canada

This paper explores the intermediate solutions between fixed prediction and forward adaptative prediction in ADPCM which consists of using a finite number, L, of preselected linear predictors of order M. The design problem of selecting the optimum set of predictors with respect to the overall prediction gain is formulated and an iterative procedure is described to obtain the solutions. The relative prediction-gain improvement is computed for a 3 sec. speech sample and for several values of L,M, and block size showing that \fraclinear{2} of the adaptative over fixed-prediction improvement in dB is reached with only L=4 and 2/3 with L=8 . The design problem solved by minimizing Itakura distance is shown to yield essentially identical performances. A linear discriminant property in the autocorrelation space is pointed out. Based on that property a pattern classification approach is proposed as an hardware-efficient coding algorithm.

关键词： Speech Encoding Autocorrelation Pattern classification Performance gain Decoding Nails linear predictive coding

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A modified adaptive transform coding scheme with post-processing-enhancement

A modified adaptive transform coding scheme with post-proces...

引用

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

作者： J. Tribolet R. Crochiere Bell Laboratories Murray Hill New Jersey Bell Laboratories Murray Hill NJ USA

The purpose of this paper is to describe a scheme for low-bit-rate transmission of speech. The scheme consists of a cascade of a codec with a post-processor. The codec is based on a modified version of the LPC Vocoder Driven Adaptive Transform coding algorithm. The Post-Processor performs a short-time Fourier analysis/synthesis at the receiver output and, by exploring the known structure of the quantization noise introduced by the codec, it is capable of speech enhancement. The performance of this scheme will be demonstrated at 9.6 kb/s and at 7.2 kb/s.

关键词： Transform coding Frequency Bandwidth Power harmonic filters linear predictive coding Vocoders Transmitters Bit rate Speech enhancement Discrete cosine transforms

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A dynamic programming approach to variable rate speech transmission

A dynamic programming approach to variable rate speech trans...

引用

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

作者： P. Papamichalis T. Barnwell School of Electrical Engineering Georgia Institute of Technology Atlanta GA USA

For many sounds, the successive frames of a speech signal do not differ significantly, and they can be represented by the same set of parameters (PARCOR coefficients). This paper examines the possibility of transmitting only some of the new parameters and replacing the rest by the parameters already transmitted. Several frames are considered, and all the possible decision sequences (paths) are examined by using a dynamic programming approach. The path which minimizes a preselected cost function is chosen for transmission, resulting in a reduced average data rate.

关键词： Dynamic programming Filters linear predictive coding Distortion measurement Speech enhancement Bit rate Cost function Prediction algorithms Pulse compression methods White noise

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