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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： J.P. Noonan P.M. Semeter Tufts University Medford MA USA Missile Systems Division Raytheon Company Tewkesbury MA UK

A data compression algorithm designed to reduce the storage requirements of a +1, -1, and 0 signal representation is proposed. The signal representation results from a fast and simple algorithm. An overview of the signal generation and compression system and details of the new compaction algorithm are presented. The results are compared to those of an existing compression technique.< >

关键词： Data compression Finite impulse response filter compression algorithms Signal processing algorithms Encoding Decoding Signal representations Frequency response Algorithm design and analysis Displays

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An evaluation of the Lempel-Ziv-Welch data compression algorithm

An evaluation of the Lempel-Ziv-Welch data compression algor...

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Southern African Conference on Communications and Signal Processing (COMSIG)

作者： H.C. Kotze G.J. Kuhn Department of Electronic and Computer Engineering University of Pretoria South Africa

The Lempel-Ziv-Welch (LZW) data compression algorithm is evaluated for use in the removal of the redundancy in computer files. The Ziv-Lempel algorithm and related algorithms are compared with respect to encoding and decoding speed, memory requirements, and compression ratio. Although the LZW algorithm is optimized for hardware implementation, the possibility of implementing it in software is considered. The most important parameter, the compression ratio, was found to be similar to or better than that of other Ziv-Lempel compression techniques, whereas coding speed and memory requirements depended on the programming techniques used.< >

关键词： Data compression Huffman coding compression algorithms Data engineering Encoding Decoding Hardware Natural languages Frequency Probability

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A nonlinear spectrum compression algorithm for the hearing impaired

A nonlinear spectrum compression algorithm for the hearing i...

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Annual Northeast Bioengineering Conference

作者： L.D. Paarmann M.D. Guiher Signal Processing Laboratory Department of Electrical and Computer Engineering Drexel University Philadelphia PA USA

A technique is proposed for increasing the speech intelligibility of persons with profound sensorineural hearing loss. The processing is based on a modification of the chirp-z transform that allows for nonlinear sampling along an arbitrary contour of the z-plane. Although the theory allows the nonlinear sampling to be arbitrary, an example showing that it is appropriate for nonlinear spectrum compression to aid the hearing-impaired is given. Samples are linearly spaced along the unit circle for a specified low-frequency segment; beyond the break-point the samples are nonlinearly spaced, perhaps spiraling inward to enhance spectral peaks, with the spacing width increasing with increasing frequency following an exponential function. A standard inverse DFT of these frequency samples returns the spectrum-compressed sequence. Processing of continuous speech requires windowing and processing of data in a manner similar to the short-time Fourier transform (STFT). The authors present the theoretical basis for, and development of, the algorithm.< >

关键词： compression algorithms Auditory system Sampling methods Speech processing Frequency Signal processing algorithms Deafness Chirp Laboratories Signal sampling

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Transform subbands coding with channel error control

Transform subbands coding with channel error control

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

作者： B. Mazor C. Hudson D. Borkowski GTE Laboratories Inc. Waltham MA USA GTE Lab. Inc. Waltham MA USA

The authors report the development of an error control scheme for the adaptive subbands excited transform coding algorithm. It is a relatively simple source-channel coding scheme which takes into account the sensitivity of the compression parameters to transmission errors. The added computational complexity, encoding-decoding delay, and rate overhead are all relatively small. The compression algorithm, with error control and without, has been implemented and tested at a total bit rate of 16 kb/s. The channel used is the memoryless binary symmetric channel with error probability up to 0.01. The results show that under error-free conditions speech quality is unchanged. However, under error conditions the quality of speech produced by the protected algorithm is consistently better.< >

关键词： Error correction Speech Programmable control Adaptive control Transform coding Computational complexity Delay compression algorithms Testing Bit rate

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Digital codec for real-time processing of broadcast quality video signals at 1.8 bits/pixel

Digital codec for real-time processing of broadcast quality ...

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1989 IEEE Global Telecommunications Conference and Exhibition 'Communications Technology for the 1990s and Beyond'

作者： M.J. Shalkhauser W.A. Whyte National Aeronautics and Space Administration Lewis Research Center Cleveland OH USA

The authors present the hardware implementation of a digital television bandwidth compression algorithm which processes standard NTSC (National Television Systems Committee) composite color television signals and produces broadcast-quality video in real time at an average of 1.8 b/pixel. The sampling rate used with this algorithm results in 768 samples over the active portion of each video line by 512 active video lines per video frame. The algorithm is based on differential pulse code modulation (DPCM), but additionally utilizes a nonadaptive predictor, nonuniform quantizer, and multilevel Huffman coder to reduce the data rate substantially below that achievable with straight DPCM. The nonadaptive predictor and multilevel Huffman coder combine to set this technique apart from prior-art DPCM encoding algorithms. The authors describe the data compression algorithm and the hardware implementation of the codec and provide performance results.< >

关键词： Codecs Digital video broadcasting Video compression TV broadcasting Hardware Pulse modulation Digital TV Bandwidth compression algorithms Color

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Real-time connected-word recognition in a noisy environment

Real-time connected-word recognition in a noisy environment

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

作者： A. Noll H.H. Hamer H. Piotrowski H.W. Ruehl S. Dobler J. Weith Philips GmbH Forschungslaboratorium Hamburg Hamburg Germany Philips Kommunikations Industrie AG Nuremberg Germany

The authors deal with connected-word recognition in high-noise environments, especially for car applications. A description of some preprocessing methods that significantly improve a given recognition system is given. A large database recorded in a car at a speed of 120 km/h (75 m.p.h.) was used for the recognition experiments. When the car database was used instead of low-noise data, the performance of the system dropped from about 1% error rate to 14.5%. Introducing a dynamic compression algorithm, a spectral weighting method, noise cancellation, and embedded training improved the performance to an error rate of about 3%.< >

关键词： Working environment noise Cognition Testing Error analysis Speech recognition Vocabulary compression algorithms Noise cancellation Telephony Government

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Techniques For The Trace-Driven Simulation Of Cache Performance

Techniques For The Trace-Driven Simulation Of Cache Performa...

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Simulation Winter Conference

作者： S.J. Eggers E.D. Lazowska Yi-Bing Lin Department of Computer Science and Engineering University of Washington Seattle WA USA

In contemporary computers, cache memories are interposed between processors and primary memories in order to decrease access time and bus traffic. Because the design of the cache is critical and the factors affecting its performance are complex, trace-driven simulation is widely used and studied. This paper surveys three interesting techniques for the trace-driven simulation of cache designs: stack analysis methodologies that make it possible to obtain performance measures for a wide variety of cache designs from a single run of the simulator, compression algorithms specifically tailored to memory reference traces, and an approach to parallel trace-driven simulation of multiprocessor caches that dramatically reduces the simulation's synchronization and thus its running time.

关键词： Computational modeling Analytical models Algorithm design and analysis Cache memory Traffic control Performance analysis Time measurement compression algorithms Computer science Design methodology

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AN ADAPTIVE REAL-TIME ECG compression ALGORITHM WITH VARIABLE THRESHOLD

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IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1988年第6期35卷 489-494页

作者： FURHT, B PEREZ, A SOFTWARE & INSTRUMENT DESIGN MIAMIFL 33152

A real-time compression algorithm has been developed which is suitable for both real-time ECG (electrocardiogram) transmission and ECG data storing. The algorithm represents a modification of the amplitude zone time epoch coding (AZTEC) technique extended with several statistical parameters used to calculate the variable threshold. The proposed algorithm has been applied in the design of a pacemaker followup system for the online ECG data transmission from the pacemaker implanted in a human being to the computer system located at the clinic.< >

关键词： Electrocardiography compression algorithms Pacemakers Algorithm design and analysis Data communication Humans

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An ECG compression algorithm for full disclosure in a solid-state real-time Holter monitor

An ECG compression algorithm for full disclosure in a solid-...

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Computers in Cardiology (CinC)

作者： N.C. Smith J.S. Platt Hewlett Packard Company McMinnville OR USA

In a real-time Holter ECG analyzer, a limited amount of solid-state memory is available for storing the 24 hours of data required for full disclosure capability. In the analyzer discussed here, 25 hours of a single channel of ECG data had to be stored in 1.5 Mbyte of RAM. Meeting the necessary data-compression goals while maintaining adequate waveform fidelity was not possible using standard data-compression techniques. The authors developed an ECG compression method for full disclosure data that meets the design requirements without imposing much of a computation burden on the real-time device. The method makes full use of the high degree of redundancy in the ECG signal. In addition, it uses filtering and subsampling techniques that, while inherently distorting, do not distort the waveform in such a way as to alter the clinical interpretation that can be made from a minaturized full disclosure report format. A quantitative analysis of the distortion is included.< >

关键词： Electrocardiography compression algorithms Solid state circuits Monitoring Data compression Erbium Manufacturing Filtering Character generation Memory management

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ASET voice coding-algorithm and implementation

ASET voice coding-algorithm and implementation

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IEEE Conference on Vehicular Technology (VTC)

作者： B. Mazor D. Veeneman D. Borkowski T. Cao GTE Laboratories Inc. Waltham MA USA GTE Lab. Inc. Waltham MA USA

The development and implementation of a medium bit-rate speech compression algorithm are reported. The algorithm, named adaptive subbands excited transform (ASET) coding, produces high-quality speech at relatively low computational complexity. Informal subjective testing of the algorithm showed no distortion or degradation in speech quality at 16 kb/s and only small degradation at 12 kb/s. In a formal mean opinion score test, 16-kb/s ASET scored 3.9/5.0 matching the score reported for standard 32-kb/s ADPCM (adaptive differential pulse code modulation). The real-time implementation is based on a single board with a single, commercially available, digital signal processing (DSP) chip. Major components of this implementation include the DSP chip, on-board data and program memories, and a dual-port memory chip. The dual-port memory is used as a buffer between I/O circuitry and the processor, allowing them to operate asynchronously. This configuration increases processing efficiency and simplifies the hardware design and operations.< >

关键词： Digital signal processing chips Signal processing algorithms Testing Degradation Pulse modulation compression algorithms Speech coding Computational complexity Code standards Modulation coding

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