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

作者： P. Poehler Junho Choi DBA Systems Inc. Melbourne FL USA

Extremely efficient compression algorithms can be devised for digital imagery data via an extension of the linear predictive coding methods utilized extensively in speech processing. When appropriately employed, these methods introduce minimal distortion. Such realizations can operate in real time, completely in the spatial domain, and are capable of reducing imagery storage requirements by an order of magnitude. This paper describes such an extension of linear predictive coding techniques for imagery data compression applications both theoretically and experimentally. This realization includes several heretofore uncombined and novel approaches to the imagery compression problem including: 2-D lattice filter prediction, adaptive quantization, and entropy coding. System implementation shows proof of feasibility and favorable performance in comparison with alternative transform techniques using the standard Minimum Mean Square Error [MMSE) fidelity criterion.

关键词： Linear predictive coding Data compression Image storage compression algorithms Digital images Speech processing Image coding Lattices Adaptive filters Quantization

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Comment on "Analysis of a Mobile Radio Communication System with Two Types of Customers and Priority"

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IEEE Transactions on Communications 1983年第11期31卷 1255-1256页

作者： K. Kawashima Musashino Electrical Communication Laboratory Nippon Telegraph and Telephone Public Corporation Musashino Tokyo Japan

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Real time implementation of a speech coding algorithm using time-domain harmonic scaling and adaptive residual coding

Real time implementation of a speech coding algorithm using ...

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

作者： A. Arora J. Melsa J. Mills Tellabs Inc. South Bend IN USA Department of Electrical Engineering University of Notre Dame Notre Dame IN USA

This paper presents a real time implementation of a speech coder at 16 kbs. The compression algorithm is a combination of two previously reported techniques: Time Domain Harmonic Scaling and Adaptive Residual Coding. Several modifications are made to reduce the hardware complexity with minimal loss in speech quality, even in the presence of channel errors at a BER of 10 -3 . A single-board, real-time, full-duplex implementation is achieved using an NEC7720 signal processing chip and an 8085 microcomputer for transmission over an RS-423 digital interface.

关键词： Speech coding Time domain analysis Signal processing algorithms Knowledge based systems compression algorithms Hardware Bit error rate Adaptive signal processing Digital signal processing chips Microcomputers

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A programmable voice digitizer using the T.I. TMS-320 microcomputer

A programmable voice digitizer using the T.I. TMS-320 microc...

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

作者： D. Daly L. Bergeron Corporate Technology Center Voice Engineering Department Wang Laboratories Inc. Lowell MA USA

With the availability of single-chip high speed digital signal processors, it is now possible to perform computationally complex voice digitization and compression within single board constraints. A new u-processor based system has been designed, within this constraint, around the TMS-320 signal processor. The hardware blocks needed to accomplish these requirements consists of a high speed microprocessor with fast multiply/accumulate, a data management section, an analog I/O section and a host DMA interface.

关键词： Microcomputers Hardware Signal processing Signal design Digital signal processing Speech processing Random access memory Telephony compression algorithms Arithmetic

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A MULTIRATE VOICE DIGITIZER BASED UPON VECTOR QUANTIZATION

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IEEE TRANSACTIONS ON COMMUNICATIONS 1982年第4期30卷 721-727页

作者： REBOLLEDO, G GRAY, RM BURG, JP TIME & SPACE PROC INC SANTA CLARACA 95051 STANFORD UNIV INFORMAT SYST LABSTANFORDCA 94305

The importance of integrating voice and data over digital networks has increased during the last few years primarily because of the growing popularity of such networks. Of particular interest are efficient voice digitizing terminals, capable of operating at various data rates in both circuit-switched and packet-switched data networks. Several such terminals, including two or more speech compression algorithms, have been proposed and implemented. Typically the terminal switches between a low-rate (500 - 4000 bits/s) vocoding scheme and a medium-rate (7000 - 16000 bits/s) waveform coding algorithm, depending on, among other things, the network congestion and on the desired voice quality and robustness. We here describe the design and simulation of a multirate voice digitizer (MRVD) that switches between two speech compression systems, each based on a recently developed vector quantization (VQ) coding technique. This technique consists of the off-line interactive design of a codebook minimizing an average distortion measure, followed by the use of the codebook in an on-line nearest neighbor encoding scheme. One of the two systems is a rate-distortion speech coder that resembles a linear predictive coding (LPC) speech compression system but has a much lower rate (800 bits/s and below). We call this the LPC-VQ system, and it is similar to other previously reported systems [15],[19],[21]. The only difference is that the LPC parameters are extracted using the Burg method instead of the autocorrelation method. We here show that this provides both qualitative and quantitative improvements. The other system of our MRVD is a residual-excited linear predictive (RELP) speech compression system using VQ in both model selection and residual digitization. The residual waveform is digitized at 1 or 2 bits/sample, resulting in rates of 7300 and 13800 bits/s, respectively. We call this the RELP-VQ system. When compared to other RELP systems [6]-[8], it is shown to have a simpler archi

关键词： Vector quantization Speech coding Linear predictive coding Switches Circuits compression algorithms Robustness Distortion measurement Nearest neighbor searches Encoding

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compression OF BLACK-WHITE IMAGES WITH ARITHMETIC CODING

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IEEE TRANSACTIONS ON COMMUNICATIONS 1981年第6期29卷 858-867页

作者： LANGDON, GG RISSANEN, J IBM Corporation San Jose CA USA

A new approach for black and white image compression is described, with which the eight CCITT test documents can be compressed in a lossless manner 20-30 percent better than with the best existing compression algorithms. The coding and the modeling aspects are treated separately. The key to these improvements is an efficient binary arithmetic code. The code is relatively simple to implement because it avoids the multiplication operation inherent in some earlier arithmetic codes. Arithmetic coding permits the compression of binary sequences where the statistics change on a bit-to-bit basis. Model statistics are studied from stationary, stationary adaptive, and nonstationary adaptive assumptions.

关键词： Image coding Arithmetic Phase detection Data communication Network address translation Biographies compression algorithms Statistics Jacobian matrices Immune system

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APPLICATION OF PSEUDO-RANDOM NOISE TO DPCM

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IEEE TRANSACTIONS ON COMMUNICATIONS 1981年第12期29卷 1763-1767页

作者： TROXEL, DE MIT DEPT ELECT ENGN & COMP SCICAMBRIDGEMA 02139

The conventional DPCM compression algorithm has been substantially improved by the substitution of a tapered, noisy quantizer for the usual tapered quantizer. This markedly reduces the visibility of defects caused by granular noise and leak. The quantizer characteristic can then be modified to achieve less slope overload, thus significantly improving image quality.

关键词： Image quality Pulse modulation Computer errors compression algorithms Noise reduction Modulation coding Redundancy Brightness Feedback loop Decoding

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16kbps Real time QMF sub-band coding implementation

16kbps Real time QMF sub-band coding implementation

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

作者： C. Galand D. Esteban Department 801 B3 IBM Laboratory La Gaude France

The availability of medium performance microprocessors, in conjunction with different concepts to generate real time efficient signal processing software on microprocessors without hardwired multipliers, allows the real time implementation of sub-band voice compression algorithms. This paper deals with the implementation of a 16kbps (8kHz, 2 bits per sample) 8 sub-band coder assuming a bank of 40 tap QMF decimator and interpolator filters, an adaptive allocation of the bits resource based on channels activity and a straigth block PCM coding of the decimated samples. The paper focuses on the implementation and software techniques which were used to achieve a complete sub-band coder with about 1.3 million of instructions per second on a processor with a 16 bits instruction and data flow.

关键词： Microprocessors Availability Signal generators Signal processing algorithms Adaptive signal processing Software performance compression algorithms Adaptive filters Channel bank filters Resource management

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Two-dimensional speech compression

Two-dimensional speech compression

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

作者： J. Alsup H. Whitehouse Naval Ocean Systems Center San Diego CA USA

The application of recent video data compression techniques to speech data is described. In order to effectively apply these techniques, the speech data should be segmented so as to achieve a high degree of correlation between corresponding samples in adjacent speech segments, allowing the formation of a two-dimensional speech "raster" with significant correlation in both dimensions. Several methods for generating such two-dimensional formats are proposed, and the results of applying the hybrid cosine-transform/DPCM compression algorithm [1] and the two-dimensional cosine transform to selected data formats are presented. Also proposed are several hardware configurations for the possible application of these results to narrowband speech processing.

关键词： TV compression algorithms Speech processing Data compression Delay Oceans Video compression Hybrid power systems Hardware Narrowband

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ANALYSIS OF SOME REDUNDANCY REMOVAL BANDWIDTH compression TECHNIQUES

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PROCEEDINGS OF THE INSTITUTE OF ELECTRICAL AND ELECTRONICS ENGINEERS 1967年第3期55卷 278-&页

作者： EHRMAN, L SIGNATRON Inc. Lexington MA USA

Three redundancy removal bandwidth compression algorithms --the floating-aperture predictor, the zero-order interpolator, and the fan interpolator--are analyzed. Theoretical expressions are found for the mean and mean-square times between output samples of these devices when the input signal is a Markov process. These expressions are evaluated for the case in which the input is a first-order Gaussian Markov process, and the resulting output sampling rates and transmission bandwidths are compared to those required by a PCM system using uniform sampling and optimum linear filter interpolation. It is shown that, given sufficient a priori knowledge of the signal process, there is little to be gained by using these redundancy removal techniques in place of the PCM system. However, if the signal statistics are unknown, the use of these algorithms instead of PCM may provide a considerable bandwidth reduction.

关键词： Bandwidth Phase change materials Signal processing Markov processes Signal sampling compression algorithms Algorithm design and analysis Nonlinear filters Interpolation Statistics

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