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A FAMILY OF SPECIAL PURPOSE MICROPROGRAMMABLE DIGITAL SIGNAL PROCESSOR ICS IN AN LPC VOCODER SYSTEM

IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1983年第1期31卷 273-281页

作者： FETTE, B HARRISON, D OLSON, D ALLEN, SP Motorola Government Electronics Group Scottsdale AZ USA

This paper discusses a family of digital signal processor integrated circuits designed to be the fundamental building blocks of an LPC vocoder, specifically LPC analysis, AMDF pitch extraction, and LPC synthesis. The IC's are custom designs intended to minimize silicon real estate for extremely high performance, novel DSP architectures, and high computational demand algorithms. Each IC shares a common architecture, that of a single internal data bus connecting input and output FIFO's, multiple arithmetic logic units, registers, and a microcode control bus. Each application is microprogrammed in register transfer language providing considerable flexibility. The CMOS design of each IC allows it to implement the high performance digital signal processing algorithms with minimal power consumption. The LPC synthesis IC allows various common filter topologies and residual excitation. The AMDF IC extracts the commonly used features for pitch and voicing. The LPC analyzer IC is microprogrammed to implement a flow form PARCOR LPC analysis. Algorithm architecture tradeoffs between computational load and memory are discussed.

关键词： Digital signal processors Digital integrated circuits linear predictive coding Vocoders Integrated circuit synthesis Computer architecture Signal processing algorithms Signal design Signal synthesis Algorithm design and analysis

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ON THE APPROXIMATION OF FIR BY IIR DIGITAL-FILTERS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1983年第1期31卷 28-34页

作者： BEDNAR, JB Exploration and Product Research Division Energy Resources Group Cities Service Company Tulsa OK USA

Kaiser's empirical formula for finite impulse response (FIR) digital filter length, as a function of transition width and rejection band loss (together with the I_{0}-\sinh window function), provides a simple FIR ... 详细信息

Kaiser's empirical formula for finite impulse response (FIR) digital filter length, as a function of transition width and rejection band loss (together with the I_linear predictive coding-\sinh window function), provides a simple FIR filter design algorithm. Recent developments in time series analysis provide a theoretical procedure for selecting the numerator and denominator orders of an infinite impulse response (IIR) digital filter whose frequency domain amplitude response is equivalent to that of a given FIR filter. In practice, this procedure has unfailingly indicated the correct denominator order, but has frequently selected a numerator order which produced a poor approximation to the desired frequency domain amplitude characteristics. Part of the problem has been due to the lack of a reasonable estimate of the quality of the approximation, and part has been due to a poor understanding of what the observed order selection criteria mean in an approximation setting. By relating the order selection algorithm to traditional methods for solving the linear predictive coding equations, this paper resolves both obstacles to minimal order approximation of FIR filters by IIR filters. As an added benefit, the relational analysis has the potential to provide an automated procedure for simultaneous order selection and IIR filter design and approximation.

关键词： Finite impulse response filter Digital filters IIR filters Approximation algorithms Optical arrays Frequency domain analysis Equations Algorithm design and analysis Time series analysis linear predictive coding

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LPC ANALYSIS-SYNTHESIS FROM SPEECH INPUTS CONTAINING QUANTIZING NOISE OR ADDITIVE WHITE NOISE

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1976年第6期24卷 488-494页

作者： SAMBUR, MR JAYANT, NS BELL TEL LABS INC MURRAY HILL NJ 07974 USA

An important problem in some communication systems is the performance of linear prediction (LPC) analysis with speech inputs that have been corrupted by (signal-correlated) quantization distortion or additive white noise. To gain a first insight into this problem, a high-quality speech sample was deliberately degraded by using various degrees (bit rates of 16 kbps and more) of differential PCM (DPCM), and delta modulation (DM) quantization, and by the introduction of additive white noise. The resulting speech samples were then analyzed to obtain the LPC control signals: pitch, gain, and the linear prediction coefficients. These control parameters were then compared to the parameters measured in the original, high quality signal. The measurements of pitch perturbations were assessed on the basis of how many points exceeded an appropriate difference limen. A distance measure proposed by Itakura was used to compare the original LPC coefficients with the coefficients measured from the degraded speech. In addition, the measured control signals were used to synthesize speech for perceptual evaluation. Results suggest that LPC analysis/synthesis is fairly immune to the degradation of DPCM quantization. The effects of DM quantization are more severe and the effects of additive white noise are the most serious.

关键词： Speech analysis Speech synthesis Additive white noise linear predictive coding Speech enhancement Quantization Signal synthesis Degradation Delta modulation Signal analysis

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MINIMUM PREDICTION RESIDUAL PRINCIPLE APPLIED TO SPEECH RECOGNITION

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1975年第1期AS23卷 67-72页

作者： ITAKURA, F NIPPON TEL & TELEGRAPH PUBL CORP ELECT COMMON LABS MUSASHINO TOKYO JAPAN BEL TEL LABS INC ACOUSTIC RES DEPT MURRAY HILL NJ 07974 USA

A computer system is described in which isolated words, spoken by a designated talker, are recognized through calculation of a minimum prediction residual. A reference pattern for each word to be recognized is stored as a time pattern of linear prediction coefficients (LPC). The total log prediction residual of an input signal is minimized by optimally registering the reference LPC onto the input autocorrelation coefficients using the dynamic programming algorithm (DP). The input signal is recognized as the reference word which produces the minimum prediction residual. A sequential decision procedure is used to reduce the amount of computation in DP. A frequency normalization with respect to the long-time spectral distribution is used to reduce effects of variations in the frequency response of telephone connections. The system has been implemented on a DDP-516 computer for the 200-word recognition experiment. The recognition rate for a designated male talker is 97.3 percent for telephone input, and the recognition time is about 22 times real time.

关键词： Speech recognition linear predictive coding Telephony Acoustic testing Acoustic measurements Power system modeling Pattern recognition Autocorrelation Dynamic programming Heuristic algorithms

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PRODUCT CODE VECTOR QUANTIZERS FOR WAVEFORM AND VOICE coding

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1984年第3期32卷 474-488页

作者： SABIN, MJ GRAY, RM Information Systems Laboratory Department of Electrical Engineering University of Stanford Stanford CA USA

Memory and computation requirements imply fundamental limitations on the quality that can be achieved in vector quantization systems used for speech waveform coding and linear predictive voice coding (LPC). One approach to reducing storage and computation requirements is to organize the set of reproduction vectors as the Cartesian product of a vector codebook describing the shape of each reproduction vector and a scalar codebook describing the gain or energy. Such shape-gain vector quantizers can be applied both to waveform coding using a quadratic-error distortion measure and to voice coding using an Itakura-Saito distortion measure. In each case, the minimum distortion reproduction vector can be found by first selecting a shape code-word, and then, based on that choice, selecting a gain codeword. Several algorithms are presented for the design of shape-gain vector quantizers based on a traning sequence of data or a probabilistic model. The algorithms are used to design shape-gain vector quantizers for both the waveform coding and voice coding application. The quantizers are simulated, and their performance is compared to that of previously reported vector quantization systems.

关键词： Product codes linear predictive coding Distortion measurement Decoding Vector quantization Shape measurement Algorithm design and analysis Speech coding Energy storage Vocoders

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A FAMILY OF SPECIAL PURPOSE MICROPROGRAMMABLE DIGITAL SIGNAL PROCESSOR ICS IN AN LPC VOCODER SYSTEM

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IEEE JOURNAL OF SOLID-STATE CIRCUITS 1983年第1期18卷 25-33页

作者： FETTE, B HARRISON, D OLSON, D ALLEN, SP Motorola Government Electronics Group Scottsdale AZ USA

This paper discusses a family of digital signal processor integrated circuits designed to be the fundamental buitding blocks of an LPC vocoder, specifically LPC analysis, AMDF pitch extraction, and LPC synthesis. The IC's are custom designs intended to minimize silicon real estate for extremely high performance, novel DSP architectures, and high computational demand algorithms. Each IC shares a common architecture, that of a single internal data bus connecting input and output FIFO's, multiple arithmetic logic units, registers, and a microcode control bus. Each application is microprogrammed in register transfer language providing considerable flexibility. The CMOS design of each IC allows it to implement the high performance digital signal processing algorithms with minimal power consumption. The LPC synthesis IC allows various common filter topologies and residual excitation. The AMDF IC extracts the commonly used features for pitch and voicing. The LPC analyzer IC is microprogrammed to implement a flow form PARCOR LPC analysis. Algorithm architecture tradeoffs between computational load and memory are discussed.

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Efficient Search and Design Procedures for Robust Multi-Stage VQ of LPC Parameters for 4 kb/s Speech coding

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第4期1卷 373-385页

作者： LeBlanc, W. P. Bhattacharya, B. Mahmoud, S. A. Cuperman, V. Carleton Univ Dept Syst & Comp Engn Ottawa ON K1S 5B6 Canada Simon Fraser Univ Sch Engn Sci Burnaby BC V5A 1S6 Canada

This paper presents a tree-searched multi-stage vector quantization scheme for LPC parameters which achieves spectral distortion lower than 1 dB with low complexity and good robustness using rates as low as 22 bits/frame. The M-L search is used and it is shown that it achieves performance close to that of the optimal search for a relatively small M. A new joint codebook design strategy for multi-stage VQ is presented which improves convergence speed and the VQ performance measures. The best performance/complexity trade-offs are obtained with relatively small size codebooks cascaded in a 3 4 stage configuration. It is shown experimentally that as the number of stages is increased above the optimal performance/complexity trade-off, the quantizer robustness and outlier performance can be improved at the expense of a slight increase in rate. Results for LAR and U P parameters are presented. A training technique that reduces outliers at the expense of a slight average performance degradation is introduced. The robustness across different languages, input spectral shapings, and in the presence of independent random channel errors is studied. Experimental results show that tree-searched multi-stage VQ significantly outperforms the split codebook approach.

关键词： Robustness linear predictive coding Speech coding Vector quantization Computational complexity Convergence Velocity measurement Noise shaping Communication channels Councils

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Pitch synchronous extended excitation in multimode CELP

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IEEE COMMUNICATIONS LETTERS 1999年第9期3卷 275-276页

作者： Tian, WS Wong, WC Law, CY Tan, AP Natl Univ Singapore Dept Elect Engn Singapore 119260 Singapore Panason Singapore Labs Pte Ltd Singapore 534415 Singapore

This letter proposes a 4-kb/s multimode code-excited linear prediction (CELP) coder with pitch synchronous extended excitation, Three modes are used for the short-term excitation, namely algebraic, extended, or stochastic excitations, together with an adaptive codebook for the long-term excitation. Comparisons with the FS-1016 and ITU-T G.723.1 coders show a performance level between these standards.

关键词： Speech Bit rate linear predictive coding Signal synthesis Stochastic processes Code standards Nonlinear filters Vectors Filtering theory Laboratories

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Efficient Complex Immittance Spectral Frequency With the Perceptual-Metric-Based Codebook Search

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IEEE SIGNAL PROCESSING LETTERS 2024年 31卷 2720-2724页

作者： Jo, Byeongho Beack, Seungkwon ETRI Daejeon 34129 South Korea

Complex-valued frequency-domain linear predictive coding (CLPC) has been developed for audio coding. Recently, representations for efficiently quantizing CLPC coefficients have been proposed, including the complex immittance spectral frequency (CISF). The CISF has limitations in that it requires signalling the sequential information to eliminate ambiguity and the highest-order coefficient (HOC) for reconstructing the CLPC coefficients. This study developed a modified CISF-based method that eliminates the need for additional information by utilizing intermediate complex polynomial properties. Furthermore, a perceptual-metric-based codebook search was proposed to improve quantization efficiency. The experimental results show robust quantization performance, while listening tests demonstrate superior audio quality compared to MPEG-D USAC long TCX at 12 kbps.

关键词： Polynomials Quantization (signal) Signal processing algorithms Noise Image reconstruction Decoding Vectors Immittance spectral frequency complex immittance spectral frequency linear predictive coding speech coding audio coding

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PRACTICAL SCHEDULING AND LINE OPTIMIZATION TECHNOLOGY FOR ASIC MANUFACTURING LINES

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IEEE TRANSACTIONS ON COMPONENTS HYBRIDS AND MANUFACTURING TECHNOLOGY 1993年第4期16卷 407-411页

作者： HASHIMOTO, C TAKEDA, T TAZAWA, S SAKURAI, T LSI Lab. NTT Atsugi Japan NTT LSI Laboratories Japan

For application specific integrated circuit (ASIC) manufacturing lines, quick product turnaround time (TAT) as well as high throughput is critical. To achieve both, efficient lot management and line operation are required. A new automatic scheduler and method of customization to increase scheduling efficiency have been developed for satisfying this requirement. Simulations with the scheduler yielded the new concept of line performance curves (LPC's). Based on these techniques, a line management algorithm that places emphasis on TAT has been developed. It employs a due date management chart (DMC) and a lot release control chart (LCC). These developments make possible dramatic improvements in the efficiency of ASIC manufacturing lines.

关键词： Application specific integrated circuits Job shop scheduling Integrated circuit technology Throughput Integrated circuit manufacture Circuit simulation Integrated circuit yield linear predictive coding Control charts Manufacturing

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