Jan H. Bons
Delft University of Technology
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Publication
Featured researches published by Jan H. Bons.
IEEE Journal on Selected Areas in Communications | 1989
Jens C. Arnbak; Jan H. Bons; J. W. Vieven
A telegraphic technique for electronic-mail correspondence between industry-standard PC facilities equipped with a writing tablet is described. Algorithms for encoding graphical information into the fixed-word format used in the common message-handling architectures are discussed. It is proposed to adopt differential chain encoding which allows an efficient representation of graphics as strings of ASCII-compatible bytes. telematic service that is feasible with this PC networking approach is compared to standard facsimile transmission of compound documents with text and graphics. >
pacific rim conference on communications, computers and signal processing | 1989
Ramjee Prasad; J.W. Vieveen; Jan H. Bons; Jens C. Arnbak
A derivation is given of the probabilities of the relative vectors which represent the approximating segments in a square coding ring for differential chain coding (DCC) of line drawings. These probabilities are required to calculate rate (coding efficiency) and distortion-versus-rate characteristics. The performances of the differential and nondifferential chain codes are compared. It is established that DCC has higher efficiency.<<ETX>>
IEEE Transactions on Broadcasting | 1994
M.J. de-Ridder-de-Groote; Ramjee Prasad; Jan H. Bons
Two new methods using an FM-radio channel for transmission of digital data to mobile terminals are examined: 1. A modification of the radio data system (RDS). In RDS, additional digital information is multiplexed with a stereo sound signal. A new system is suggested where the data signal can be multiplexed with a mono audio signal. This causes extension to the bandwidth available for the data signal, and therefore the RDS bitrate can be increased. Error calculations are performed both for the original RDS system and for the new system. 2. Orthogonal frequency division multiplexing (OFDM). OFDM is used in the digital audio broadcasting system (DAB), which is designed to transmit digital audio in the FM band. In OFDM a signal is divided over a large number of 2- or 4-PSK modulated orthogonal subcarriers. The subcarriers of 6 different programmes are multiplexed in one beam to reduce the effects of frequency selectivity of the transmission channel. A new system based on OFDM is proposed, in which the carriers of each programme are transmitted in one FM-channel with a bandwidth of 200 kHz instead of multiplexed with the carriers of other programmes. Error calculations are performed for the subcarriers used in the OFDM modulation method. >
ieee region 10 conference | 1990
Kun Liu; Jan H. Bons; Jens C. Arnbak
The authors present a system for combining the two related yet separated modes of electronic document transfer, namely, electronic-mail (E-mail) and facsimile. Messages prepared at desk-top terminals, supporting compound document creation and editing, and submitted to the E-mail network can thus be received by both interconnected terminals in the common E-mail architecture and by facsimile terminals connected to the public switched telephone networks (PSTN). The latter transmission is achieved by facsimile gateways which connect the two different architectures and perform the necessary protocol conversions.<<ETX>>
vehicular technology conference | 1993
Ramjee Prasad; Paul A D Spaargaren; Jan H. Bons
A system is proposed for informing passengers who travel by train during their journey. An important part of the system is mobile communication. Teletext is chosen as an information medium, and the possibility of mobile teletext reception is analyzed considering selection diversity (SD) and maximal ratio combining (MRC). The bit error probability is calculated by modeling the channel as a shadowed-Rician fading channel. Picture quality is measured with average number of errors per page (ANEP). Thus performance is measured in terms of bit error probability (BER) and average number of errors per page. Finally, the level crossing rate, average fade, and average nonfade durations are also evaluated. >
IEEE Transactions on Broadcasting | 1992
Ramjee Prasad; Richa Joshi; Jan H. Bons
Four parameters are defined to measure the performance of a teletext system, namely, the probability of delivering a message within specified time, the transmission efficiency, the average number of errors per page, and the throughput. Each of them is derived and computational results are presented taking the UK teletext system as an example. These parameters are compared in their ability to characterize the performance of the teletext system. >
IEEE Transactions on Image Processing | 1996
George Muller; Christiane Kloditz; Jan H. Bons; Ramjee Prasad
This paper presents a method to apply progressive transmission to line drawings using the wavelet transform. Experiments have been conducted and showed that the wavelet transform, combined with a quantization step, performs progressive transmission using a data rate comparable to standard chain coding at the expense of almost no visually perceptible distortion.
IEEE Transactions on Broadcasting | 1992
Hans Boeve; Ramjee Prasad; Jan H. Bons
The introduction of double-bundle Reed-Solomon (RS) error correcting codes in UK teletext is proposed. In order to implement this, one or two rows of parity-check bytes must be added. Single-bundle and double-bundle codes are defined. The performance of the UK teletext system is evaluated, when rows of parity-check bytes in the bundles are used and the computer simulations are described. A software written coder and decoder are described which have been tested in a laboratory environment. Results obtained from simulations and implementations are in good agreement with analytical results. The performance of the teletext system is found to be enhanced by the introduction of the RS codes. >
transactions on emerging telecommunications technologies | 1993
Ronald C. N. Koch; Ramjee Prasad; Jan H. Bons
In store-and-forward and multi-media speech applications, e.g. in voice-mail or when adding verbal comments to a document, the speech encoder can be allowed more time and higher complexity than in real-time speech coding. Decoding however must be simple when a low-cost terminal (e.g., a PC) is used. This kind of coding algorithm is called an asymmetric algorithm. Asymmetric algorithms exploit encoding delay to code the signal efficiently. An existing asymmetric coding algorithm is Vector Quantization (VQ). A new asymmetric algorithm is proposed, based on the coding of the time intervals between zero-crossings (frames) using VQ and silence deletion. The algorithm takes advantage of the empirical fact that infinitely clipped speech retains its intelligibility. The algorithm has been implemented using software and its performance was evaluated.
Electronics Letters | 1991
Ramjee Prasad; Jan H. Bons; P.A.D. Spaargaren