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PREDICTIVE ANALOG-TO-DIGITAL CONVERSION OF DOPPLER ULTRASOUND SIGNALS

IEEE TRANSACTIONS ON BIOMEDICAL ENGINEERING 1995年第3期42卷 260-268页

作者： BOE, S KRISTOFFERSEN, K VINGMED SOUND AS N-1324 LYSAKER NORWAY

Analog to digital conversion in multigate Doppler ultrasound systems for blood velocity measurements is a technological challenge, The echoes must be digitized at a rate determined by the system bandwidth (typically 2 MHz), and the dynamic range is large (16 bits or more) due to the presence of strong, low-frequency Doppler clutter echoes originating from slowly moving tissue, Off-the-shelf A/D converters do not meet these requirements with the transducer configuration employed by contemporary Doppler systems, Analysis reveals a 5-b reduction in required wordlength for an AID converter in a predictive feedback loop when the maximum clutter frequency is about 1.5% of the pulse repetition frequency, The prediction error filter is recursive. Alternatively, first- and second-order DPCM (differential pulse code modulation) yield 4 and 6 b, respectively. With short input segments (from a high-resolution Color Flow Mapper), the results are, in the above order, 4, 4, and 5 b, The results are verified by processing an experimental Doppler signal.

关键词： Doppler Ultrasonography Error of prediction system bandwidth Speed measurement differential pulse code modulation clutter Echoes digital conversion Analog-digital conversion Feedback loop Doppler signal Dynamic range

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REAL-TIME VIDEO COMPRESSION USING differential VECTOR QUANTIZATION

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS FOR VIDEO TECHNOLOGY 1995年第1期5卷 14-24页

作者： FOWLER, JE ADKINS, KC BIBYK, SB AHALT, SC Department of Electrical Engineering Ohio State Uinversity Columbus OH USA

This paper describes hardware that has been built to compress video in real time using full-search vector quantization (VQ). This architecture implements a differential-vector-quantization (DVQ) algorithm and features a special-purpose digital associative memory, the VAMPIRE chip, which has been fabricated in 2 mum CMOS. We describe the DVQ algorithm, its adaptations for sampled NTSC composite-color video, and details of its hardware implementation. We conclude by presenting both numerical results and images drawn from real-time operation of the DVQ hardware.

关键词： CMOS memory circuits associative processing content-addressable storage differential pulse code modulation real-time systems vector quantisation video coding CMOS DPCM VAMPIRE chip architecture differential vector quantization full-search vector quantiza Video coding ARCHITECTURE REAL TIME OPERATION differential pulse code modulation Content addressable storage Building construction Associative Processes real-time systems Vector quantization CMOS memory circuits

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VLSI ARCHITECTURE FOR HDTV SUBBAND CODING USING GQMFS FILTERBANKS

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1995年第1期41卷 1-11页

作者： ALASMARI, AK AHMED, RE Department of Electrical Engineering King Saud University Riyadh Saudi Arabia

A VLSI architecture for a low complexity intraframe subband image coding for HDTV signals is presented. The Generalized Quadrature Mirror Filters (GQMFs), which have smaller overall delay, are optimized in order to achieve high coding efficiency. The filter design exploits a symmetry property among different filter coefficients which, in turn, reduces the hardware complexity of the VLSI architecture substantially. The architecture is designed with 1 mu m CMOS technology using the scalable design rules of MOS Integrated Services Inc. (MOSIS). The VLSI architecture contains approximately 200,000 transistors and it is capable of operating at a speed of 34 MHz. The HDTV signals are decomposed into seven bands. The baseband is DPCM encoded and the high bands are PCM encoded. It is shown that high quality HDTV images can be obtained at as low bit rate as 0.654 bpp (bits/pixel).

关键词： subband coding ARCHITECTURE differential pulse code modulation Filter coefficient Building construction Very large scale integration filter design

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DETECTION AND CORRECTION OF TRANSMISSION ERRORS IN DPCM IMAGES

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS FOR VIDEO TECHNOLOGY 1995年第2期5卷 166-171页

作者： CHANG, CH LEOU, JJ NATL CHUNG CHENG UNIV INST COMP SCI & INFORMAT ENGNCHIAYI 621TAIWAN

In this paper, a new approach to detection and correction of transmission errors in DPCM images is proposed. Here each transmission error (streak noise) is detected via a sequence of the WMW rank-sum tests performing on a sliding window using four particularly designed grouping schemes. Instead of four possible patterns used in Kundu and Wu, in the proposed approach 18 possible detection patterns represented as a decision tree are identified. And the transmission error detection procedure is equivalent to tracing the decision tree. The estimated gray level shift of a detected streak noise is used to locate the streak origin. Then instead of using previous line replacement, each streak noise can be almost deleted by compensating the corrupted pixels with the estimated gray level shift. Experimental results show the feasibility of the proposed approach.

关键词： sliding windows differential pulse code modulation gray level Decision Trees Transmission errors

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DPCM PICTURE TRANSMISSION OVER NOISY CHANNELS WITH THE AID OF A MARKOV MODEL

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IEEE TRANSACTIONS ON IMAGE PROCESSING 1995年第11期4卷 1473-1481页

作者： EMAMI, S MILLER, SL UNIV FLORIDA DEPT ELECT ENGN GAINESVILLE FL 32611 USA

The problem of DPCM picture transmission over noisy channels is considered, It is well known that DPCM systems are very sensitive to channel errors, The goal in this work is to build robustness against channel errors, Three methods are proposed in this paper and are obtained by modeling the encoded signal as a Markov sequence, First, an optimum method for decoding correlated sequences is derived, and it is shown to require Viterbi decoding, Then, a modified MAP method (MMAP) for Markov sequences is described, A maximal signal-to-noise (MSNR) receiver for DPCM systems is also developed that minimizes the distortion power due to channel errors, The appropriate cost matrix for this receiver is computed, These methods are applied to DPCM picture transmission over noisy channels and are compared with a recent method, The SNR graphs, as well as subjective examination of the received pictures, demonstrate that the proposed procedures are quite effective and superior to that method. Among the proposed methods, the MSNR receiver was found to be more effective than the others for a given order of the Markov model. It is observed that the proposed methods are most beneficial for low detail pictures.

关键词： differential pulse code modulation noisy channel distortion power picture Receivers Markov Markov models Automatic incident detection

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PROGRESSIVE DPCM SYSTEM WITH BLOCK TRUNCATION CODING

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ELECTRONICS LETTERS 1995年第21期31卷 1821-1822页

作者： KUO, CH CHEN, CF Department of Electrical Engineering Tatung Institute of Technology Taipei Republic of China

A progressive DPCM system with a block truncation coding technique for encoding the prediction error is proposed. A classified progressive DPCM scheme is also proposed. to make the progressive DPCM system reduce the bit rate more efficiently under the limited bit rate. The simulation results-indicate that the proposed method achieves excellent image quality under 2 bit/pixel and is superior to other proposed methods.

关键词： BLOCK codeS IMAGE CODING differential pulse code modulation

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From specification to implementation-the application of high-performance tools to the design of a digital filter for DECT

From specification to implementation-the application of high...

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IEE 15th SARAGA Colloquium on Digital and Analogue Filters and Filtering Systems

作者： P. Willekens D. Devisch Mentor Graphics EDC Belgium Mentor Graphics Japan

In this paper a case-study is presented describing the design of a digital recursive filter using DSP Station from Mentor Graphics. The designed filter is part of an ADPCM Transcoder IC for the DECT (Digital European Cordless Telephone) standard. This IC includes on-board sigma-delta AD/DA converters. The paper shows the design of the digital filter starting from a frequency domain specification using tools for filter synthesis, frequency domain simulation, coefficient optimisation, finite wordlength effect analysis and high-level synthesis. A comparison between a standard biquad structure and a wave digital filter (WDF) structure is made. The resulting bitserial RTL level netlist generated by the high-level synthesis tool, Mistral, can be used for FPGA, standard cell or gate array implementation.

关键词： Cordless telephone systems differential pulse code modulation Frequency domain synthesis Circuit optimization High-level synthesis Logic arrays Recursive digital filters

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An efficient embedded ADPCM coder

An efficient embedded ADPCM coder

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Fifth IEE Conference on Telecommunications 1995

作者： S.D. Zhang G.B. Lockhart University of Leeds UK

Embedded speech coding offers a graceful and acceptable degradation of speech quality when a network or channel becomes congested and transmitted information is lost. Unlike PCM which is inherently an embedded code, the more efficient differential codes such as DPCM and delta modulation (DM) do not possess embedded coding properties without modification. Embedded versions of both DPCM and DM have therefore been proposed (Goodman (1980), Wassell et al. (1988), Jayant (1983), Zhang and Lockhart (1991)). The present authors demonstrate a novel embedded adaptive DPCM (EADPCM) speech coder. The coder uses the explicit noise coding technique (Jayant) to generate a supplemental bit stream which may be wholly or partly discarded during transmission. A new low-complexity algorithm is employed to match the supplemental adaptive PCM (APCM) coder to the ADPCM reconstruction error. The step size for the APCM coder is determined in two stages. First, a basic step size is estimated from the ADPCM step based on instantaneous or syllabic adaptation or their combination. This is followed by further instantaneous step size adjustment based on the APCM output. Simulation shows that a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 (or 40) kb/s to 16 kb/s in 8 kb/s steps.< >

关键词： differential pulse code modulation Speech coding

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Compression of integral 3D TV pictures

Compression of integral 3D TV pictures

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Fifth International Conference on Image Processing and its Applications, 1995.

作者： M. Forman A. Aggoun M. McCormick De Montfort University UK

The development of a compression algorithm for reduction of the transmission bit rate of 3D integral TV pictures is discussed. The proposed full parallax 3D image compression algorithm takes advantage of the cross correlation between the multiple images recorded on a CCD placed behind a directionally selective microlens array as well as the correlation inherent within each image. Tests carried out show that by using a hybrid DPCM/DCT coding scheme to code a still 3D integral TV image, a high compression performance can be obtained while retaining all the necessary information to reconstruct a faithful duplicate of the original 3D image. A coding scheme for moving integral 3D images is also described.< >

关键词： Charge coupled image sensors Correlation Data compression differential pulse code modulation discrete cosine transforms Image reconstruction Stereo vision Transform coding Video coding

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SHARED TIME-DIVISION DUPLEXING - AN APPROACH TO LOW-DELAY HIGH-QUALITY WIRELESS DIGITAL SPEECH COMMUNICATIONS

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1994年第4期43卷 934-945页

作者： WONG, WC SUNDBERG, CEW SESHADRI, N AT&T BELL LABS SIGNAL PROC RES DEPTTECH STAFFMURRAY HILLNJ 07974

Various strategies to provide low-delay high-quality digital speech communications in a high-capacity wireless network are examined. Various multiple access schemes based on time-division and packet reservation are compared in terms of their statistical multiplexing capabilities, sensitivity to speech packet dropping, delay, robustness to lossy packet environments, and overhead efficiency. In particular, a low-delay multiple access scheme, called shared time-division duplexing (STDD) is proposed. This scheme allows both the uplink and downlink traffic to share a common channel, thereby achieving high statistical multiplexing gain even with a low population of simultaneous conversations. We also propose a choice of low delay, high quality speech coding and digital modulation systems based on adaptive DPCM, with QDPSK or pseudo-analog transmission (skewed DPSK), for use in conjunction with the STDD multiple access protocol. The choice of the alternative systems depends on required end-to-end delay, recovered speech quality and bandwidth efficiency. Typically, with a total capacity of 1 MBaud, 2 ms frame and 8 kBaud speech coding rate, low delay STDD is able to support 48 pairs of users compared to 38, 35, and 16 for TDMA with speech activity detection, basic TDMA and PRMA respectively. This corresponds to respective gains of 26%, 37% and 200%.

关键词： delays differential phase shift keying differential pulse code modulation digital radio packet radio networks packet reservation multiple access personal communication networks speech coding telephone traffic time division multiple access QDPSK STDD adap differential pulse code modulation Packet radio networks speech coding Telephone service packet reservation multiple access differential phase shift keying differential quadrature phase shift keying time division multiple access FYB gene Digital radio Personal communication networks Time division

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