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EFFECTS OF INPUT QUANTIZATION IN FLOATING-POINT DIGITAL PULSE-COMPRESSION

IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1985年第2期33卷 434-435页

作者： ARIDGIDES, A MORGAN, DR Electronics Laboratory General Electric Company Limited Syracuse NY USA

A computer simulation of an FFT-based digital LFM pulse compressor using vector floating point arithmetic is presented, showing the effects of retaining various number of mantissa bits at the input quantizer. Plots of the compressed pulse waveforms and rms and peak sidelobe levels as a function of the number of mantissa bits are shown.

关键词： Quantization Pulse compression methods Speech processing linear predictive coding Bit rate Signal processing algorithms Encoding Signal to noise ratio Signal processing Speech coding

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ON THE ACCURACY OF THE KUMARESAN-TUFTS METHOD FOR ESTIMATING COMPLEX DAMPED EXPONENTIALS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1987年第2期35卷 231-235页

作者： PORAT, B FRIEDLANDER, B SAXPY COMP CORP SUNNYVALECA 94086

Recently, Kumaresan and Tufts (KT) presented a method for estimating the parameters of damped exponential waveforms in additive white noise. The KT method uses singular value decomposition (SVD) of the data matrix, with truncation and backward prediction to improve the accuracy of the estimates. The KT method was demonstrated to have a very good performance, in comparison to traditional methods used for the same problem (e.g., Prony's method). Kumaresan and Tufts also showed, by numerical simulations, that the variances of the estimates obtained by their method approaches the Cramer-Rao lower bounds for selected test cases. In this correspondence, we provide a quantitative accuracy analysis of the KT method. The analysis is based on first-order Taylor series approximations of the estimated parameters around their true values. No assumptions are made on the number of data points used, but it is assumed that the noise level is small enough for the first-order approximations to be valid. Results of the analysis are illustrated by some numerical examples. These results confirm the good performance of the KT method, and show the effect of the user-chosen parameters on the accuracy of the estimates.

关键词： Signal processing algorithms linear predictive coding Speech synthesis Parameter estimation Lattices Speech analysis Algorithm design and analysis Speech processing Equations Acoustic measurements

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Adaptive predictive coding with transform domain quantization using block size adaptation and high-resolution spectral modeling

Adaptive predictive coding with transform domain quantizatio...

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IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： B.R.U. Bhaskar COMSAT Laboratories Clarksburg MD USA

The adaptive predictive coding with transform domain quantization (APC-TQ) technique was proposed by Bhaskar (1991) for the compression of audio signals. Since then, significant developments have taken place leading to a reduction in the coding rate. While enhancing the audio quality. These developments include (i) the use of block size adaptation to exploit the variations in the stationarity of the signal, (ii) high resolution spectral modeling using LPC analysis orders up to 64, and (iii) an adaptive bit-allocation procedure to minimize coding noise power as well as minimize the perception of coding noise. The result is a near transparent quality compression of 5 kHz bandwidth audio at a rate of 17 kbit/s. This technology will find applications in the distribution and transmission of AM quality audio programming over low rate channels such as the INMARSAT Standard A, B and aeronautical systems.< >

关键词： predictive coding Quantization Discrete transforms Signal resolution linear predictive coding Bandwidth Bit rate Humans predictive models Laboratories

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Speech Recognition Using Speaker Adaptation by System Parameter Transformation

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第1期2卷 63-68页

作者： Hao, Ying Fang, Ditang Texas A&M Univ Dept Comp Sci College Stn TX 77843 USA Tsinghua Univ Dept Comp Sci & Technol Beijing 100084 Peoples R China

This investigation presents a speaker adaptation scheme which transforms the prototype speaker's Hidden Markov word models to those of a new speaker. Transformations are applied to both the state transition matrix and the probability distribution functions of a Hidden Markov word model. These transformations are optimized through maximizing the joint probability of a set of input pronunciations of the new speaker. Details of these parameter transformations and experimental verification are presented. The test uses a 210-word vocabulary with each having a four-state Hidden Markov word model. The test speaker consists of three males and two females with one male heavily accented. By having the system retrained up to a four-minute adaptation speech, a subset of the 210-word vocabulary, the performance shows an improvement of recognition accuracy from 22.5% to 92.1%.

关键词： Speech recognition Hidden Markov models Signal processing Prototypes Testing Vocabulary Speech processing Probability distribution Computer science linear predictive coding

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MICROPROCESSOR REALIZATION OF A linear predictive VOCODER

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1977年第5期25卷 379-387页

作者： HOFSTETTER, EM TIERNEY, J WHEELER, O MIT LINCOLN LAB LEXINGTON MA 02173 USA

A microprocessor realization for a linear predictive vocoder is presented. The goal was a low-power, low-cost, compact special-purpose realization of a narrow-band speech terminal. The resultant design is a general-purpose two-bus structure running at a 150 ns cycle time, using as the basic signal processing element, four of the AMD 2901 CPE chips. This basic structure is augmented by a four-cycle multiplier to allow for sufficient signal processing power. The design concessions that mark the linear predictive coding microprocessor (LPCM) as a special-purpose machine designed to be a speech terminal are: limited I/O and limited memory. The present design requires 162 dual-in-line packages, dissipates less than 45 W and occupies about \fraclinear predictive codinglinear predictive coding ft 3 .

关键词： Microprocessors Vocoders linear predictive coding Narrowband Signal processing Speech coding Speech analysis Modems Signal processing algorithms Signal design

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A REAL-TIME LPC-BASED VOCAL-TRACT AREA DISPLAY FOR VOICE DEVELOPMENT

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JOURNAL OF VOICE 1994年第4期8卷 314-319页

作者： ROSSITER, D HOWARD, DM DOWNES, M UNIV YORK DEPT ELECTRSIGNAL PROC VOICE & HEARING RES GRPYORK YO1 5DDN YORKSHIREENGLAND

This article reports the design and implementation of a graphical display that presents an approximation to vocal tract area in real time for voiced vowel articulation. The acoustic signal is digitally sampled by the system. From these data a set of reflection coefficients is derived using linear predictive coding. A matrix of area coefficients is then determined that approximates the vocal tract area of the user. From this information a graphical display is then generated. The complete cycle of analysis and display is repeated at approximately 20 times/s. Synchronised audio and visual sequences can be recorded and used as dynamic targets for articulatory development. Use of the system is illustrated by diagrams of system output for spoken cardinal vowels and for vowels sung in a trained and untrained style.

关键词： VOICE TRAINING VISUAL DISPLAYS linear predictive coding

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TEXT-TO-SPEECH USING LPC ALLOPHONE STRINGING

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1981年第2期27卷 144-152页

作者： LIN, KS GOUDIE, KM FRANTZ, GA BRANTINGHAM, GL Texas Instruments Inc. USA

A low cost voice response system is presented, which performs text-to-speech conversion of any English text. The system is built around an LPC synthesizer chip and a microprocessor. Text-to-allophone rules are used to convert an input string of ASCII characters into allophonic codes. LPC parameters are then drawn from an allophone library, which takes very little storage space, and concatenated using a fast and simple algorithm to produce natural sounding speech.

关键词： Speech synthesis linear predictive coding Synthesizers Costs Libraries Lattices Vocabulary Microprocessors Filters Instruments

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VOICE coding AND TREE ENcoding SPEECH COMPRESSION SYSTEMS BASED UPON INVERSE FILTER MATCHING

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IEEE TRANSACTIONS ON COMMUNICATIONS 1982年第4期30卷 711-720页

作者： MATSUYAMA, Y GRAY, RM STANFORD UNIV INFORMAT SYST LABSTANFORDCA 94305

Two speech compression systems based on codebooks of inverse filters produced by off-line linear predictive coding (LPC) and vector quantization (VQ) techniques are considered. The first system is a pitch excited vocoder that is a variation on a speech coding system based upon vector quantization. The encoder selects an LPC reverse filter from a finite codebook that best "matches" an observed frame of sampled speech. This filter is in turn used to determine the voicing and digitized pitch information. Unlike LPC systems, the digitization is performed in a single step on the data rather than separate modeling and digitization steps. The second system is a tree encoding system that uses the filter selected by an inverse filter matching vocoder to "color" a tree that is then searched for a minimum distortion path for the original sampled speech waveform. This system can be viewed as a hybrid between an adaptive predictive coder and a universal tree encoder. The two systems are described, simulated, and compared with other similar systems.

关键词： Encoding Speech coding Matched filters linear predictive coding Vector quantization Vocoders Nonlinear filters Information filtering Information filters Predistortion

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FEATURE-EXTRACTION BY SYSTEM-IDENTIFICATION

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IEEE TRANSACTIONS ON SYSTEMS MAN AND CYBERNETICS 1982年第1期12卷 42-50页

作者： EISENSTEIN, BA VACCARO, RJ PRINCETON UNIV DEPT ELECT ENGN PRINCETON NJ 08540 USA

A method for the extraction of features for pattern recognition by system identification is presented. A test waveform is associated with a parameterized process model (PM) which is an inverse filter. The structure of the PM corresponds to the redundant information in a waveform, and the parameter values correspond to the discriminatory information. The PM used in this research is a linear predictive system whose parameters are the linear predictive coefficients (LPC's). This technique is applied to feature extraction of electrocardiograms (ECG's) for differential diagnosis. The LPC's are calculated for each ECG and used as a feature vector in a hypergeometric affine N-space spanned by the LPC's. The efficacy of this feature extraction technique is tested by three different perturbation methods, namely noise, matrix distortion, and a newly developed method called directed distortion. Both the Euclidean and Itakura distances between feature vectors in N-space are shown in increase with increasing perturbation of the template waveform. The monotonic behavior of a distance measure is a necessary attribute of a valid feature space. Thus the perturbation analyses done in this research verify the viability of using the parameters of a process model as a feature vector in a pattern recognition scheme.

关键词： Feature extraction System identification linear predictive coding Pattern recognition Testing Electrocardiography Data mining Filters Vectors Perturbation methods

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A SINGLE CHIP SPEECH SYNTHESIZER USING A SWITCHED-CAPACITOR MULTIPLIER

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1983年第1期31卷 313-323页

作者： GREGORIAN, R AMIR, G UNIVERSAL SEMICOND INC SAN JOSECA 95112

A single chip speech synthesizer was designed using a switched-capacitor multiplier to implement the LPC algorithm. The chip contains the LPC-10 filter, 20 kbit ROM, all control logic, a three-pole switched-capacitor low-pass filter, and an audio amplifier capable of driving a speaker directly. The chip was fabricated in 5 μm CMOS technology and is 218 mils on the side.

关键词： Speech synthesis Synthesizers linear predictive coding Speech analysis CMOS technology Read only memory Low pass filters Bit rate Silicon Resonator filters

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