RS（16,12）缩短码编译码原理及性能分析

在各类数字通信系统以及计算机存储和运算系统经常利用差错控制编码降低误码率,提高通信质量,满足对数据传输通道可靠性的要求。RS码是一种性能优良的前向纠错码,具有同时纠正随机错误和突发错误的能力,它的构造特点决定了其非常适合于纠正突发性错误。文中在阐述RS系统码编译码原理的基础上,提出了RS（16,12）缩短码的编译码方法,利用MATLAB对R S（16,12）缩短码在高斯信道和瑞利信道条件下的纠错能力进行仿真,并分析其纠错性能。     

The Loop Coverage Comparison Between TCM and Echo Canceller Under Various Noise Considerations

This paper describes loop coverage evaluations of two-wire digital subscriber transmission systems to support the CCITT recommended 1-430 interface. Several range-limiting factors are discussed and a cable utility design for two representative two-wire digital subscriber transmission systems is presented. The two transmission systems are time compression multiplexing (TCM) and echo canceller (EC), utilizing the AMI line code and full cosine rolloff line equalization for both systems. A new impulse noise evaluation is applied to cable utility design, because impulse noise is one of the most stringent range-limiting factors, especially for TCM in utilizing existing subscriber loops. Comparison of TCM with EC is also carried out according to various range-limiting factors. As a result, it is noted that the 320 kbit/s line bit rate TCM is more applicable in NTT's exchange area coverage than the 160 kbit/s EC to achieve the2B + Dbasic interface transmission.     

Digital optical CATV system using hubbed distribution architecture

A hubbed distribution architecture based on a bus-star configuration, three-level single single-mode fiber transmission using M-TPC line code, and high-speed IC technology is proposed. The architecture is cost-effective and compact. It is well suited to metropolitan area CATV systems and would allow a gradual transition to a broadband ISDN (integrated-services digital network) subscriber loop in the future. The proposed architecture is applied to a digital optical CATV system prototype. The equipment making up this system consists of a transmitter, hubs including video selectors, and subscriber unit. Two video channels are selected for each subscriber from eight video channels at hub stations. The transmission speed is 900 Mb/s, three-level for the trunk line and 200 Mb/s, three-level for each subscriber loop. Very compact and low-cost equipment is realized by using recently developed high-speed ICs. Experimental results show that the system satisfies requirements and that its commercial application is feasible     

Enhanced Low Complex Double Error Correction Coding with Crosstalk Avoidance for Reliable On-Chip Interconnection Link

As the technology scales down, shrinking geometry and layout dimension, on- chip interconnects are exposed to different noise sources such as crosstalk coupling, supply voltage fluctuation and temperature variation that cause random and burst errors. These errors affect the reliability of the on-chip interconnects. Hence, error correction codes integrated with noise reduction techniques are incorporated to make the on-chip interconnects robust against errors. The proposed error correction code uses triplication error correction scheme as crosstalk avoidance code (CAC) and a parity bit is added to it to enhance the error correction capability. The proposed error correction code corrects all the error patterns of one bit error, two bit errors. The proposed code also corrects 7 out of 10 possible three bit error patterns and detects burst errors of three. Hybrid Automatic Repeat Request (HARQ) system is employed when burst errors of three occurs. The performance of the proposed codec is evaluated for residual flit error rate, codec area, power, delay, average flit latency and link energy consumption. The proposed codec achieves four magnitude order of low residual flit error rate and link energy minimization of over 53 % compared to other existing error correction schemes. Besides the low residual flit error rate, and link energy minimization, the proposed codec also achieves up to 4.2 % less area and up to 6 % less codec power consumption compared to other error correction codes. The less codec area, codec power consumption, low link energy and low residual flit error rate make the proposed code appropriate for on chip interconnection link.     

Optical broadband access using ATM on a passive optical network

Within the framework of the ‘FITL (fibre in the loop) evolution’, Siemens is developing a prototype based on a passive optical network (PON) for the economic provision of BISDN services in the subscriber access area. It provides 32 subscribers with BISDN access up to a maximum user data rate of 45 Mbit/s per subscriber. The information is transferred via ATM cells and PON-specific overhead in both directions on the PON at a data rate of 155.52 Mbit/s. For transmission to the subscribers, the time division multiplex procedure (TDM) is used. For transmission in the opposite direction, a highly flexible time division multiple access procedure (TDMA) controls the access to PON for the subscriber-side line terminations. In order to keep the transmission system inexpensive emphasis is placed on a digital calibration of the fibre loop delay, a burst transmitter at the subscriber side without amplitude recovery and a burst receiver at the exchange side with purely digital bit synchronization. However, to achieve these features a small part of data rate must be used as a PON specific overhead.     

An integrated error correction and detection system for digitalaudio broadcasting

Hybrid in-band on-channel digital audio broadcasting systems deliver digital audio signals in such a way that is backward compatible with existing analog FM transmission. We present a channel error correction and detection system that is well-suited for use with audio source coders, such as the so-called perceptual audio coder (PAC), that have error concealment/mitigation capabilities. Such error mitigation is quite beneficial for high quality audio signals. The proposed system involves an outer cyclic redundancy check (CRC) code that is concatenated with an inner convolutional code. The outer CRC code is used for error detection, providing flags to trigger the error mitigation routines of the audio decoder. The inner convolutional code consists of so-called complementary punctured-pair convolutional codes, which are specifically tailored to combat the unique adjacent channel interference characteristics of the FM band. We introduce a novel decoding method based on the so-called list Viterbi algorithm (LVA). This LVA-based decoding method, which may be viewed as a type of joint or integrated error correction and detection, exploits the concatenated structure of the channel code to provide enhanced decoding performance relative to decoding methods based on the conventional Viterbi algorithm (VA). We also present results of informal listening tests and other simulations on the Gaussian channel. These results include the preferred length of the outer CRC code for 96-kb/s audio coding and demonstrate that LVA-based decoding can significantly reduce the error flag rate relative to conventional VA-based decoding, resulting in dramatically improved decoded audio quality. Finally, we propose a number of methods for screening undetected errors in the audio domain     

Analysis of video transmission over lossy channels

A theoretical analysis of the overall mean squared error (MSE) in hybrid video coding is presented for the case of error prone transmission. Our model covers the complete transmission system including the rate-distortion performance of the video encoder, forward error correction, interleaving, and the effect of error concealment and interframe error propagation at the video decoder. The channel model used is a 2-state Markov model describing burst errors on the symbol level. Reed-Solomon codes are used for forward error correction. Extensive simulation results using an H.263 video codec are provided for verification. Using the model, the optimal tradeoff between INTRA and INTER coding as well as the optimal channel code rate can be determined for given channel parameters by minimizing the expected MSE at the decoder. The main focus of this paper is to show the accuracy of the derived analytical model and its applicability to the analysis and optimization of an entire video transmission system     

A TDMA satellite communication system for ISDN services

The authors propose a TDMA (time-division multiple-access) satellite communication system suitable for both basic and primary ISDN services in addition to B-channel transmission lines among ISDN switches. To provide these services economically and efficiently, the proposed system uses an offset quadrature phase-shift keying (QPSK) burst model coupled with high-coding gain and rate-forward-error correction, demand assignment technology for each data burst, and a transmission and reception transponder (carrier) hopping scheme. To realize cost-effective transmission and subscriber lines and to increase the reliability of TDMA terminals, the major functions of TDMA terminals have been implemented in large-scale integrated circuits (LSICs), monolithic ICs (MICs) and hybrid ICs (HICs). The experimental results show narrow spectrum transmission with little bit error probability degradation for the offset QPSK burst modem and perfect operation of demand assignment and carrier hopping capability of the system     

A Combined Interleaving Technique for Trellis Coded MPSK Systems in Rayleigh Fading Channels

Freedom of mobility is the latesttrend in the evolution of digital communicationsystems. In these systems bandwidth and powerconstraints limit the mobility of the users. TCM(trellis coded modulation) is a combined coding andmodulation scheme which increases the productivity ofa digital communication system without increasing thetransmitted power or the required bandwidth. A mobilecommunication channel is affected from noise andfading because of the multipath propagation. On aRayleigh fading channel, coding should be used withtechniques which uncorrelate the received energy ofeach consecutive coded symbol. Interleaving is one ofthese techniques and used to make burst errors intorandom errors which can be corrected by errorcorrecting codes. In this paper, the trellis codingsystem which combines a regular convolutional encoder,bit interleaver, coordinate interleaver, and anappropriate decoder is considered. The analytical biterror probability upper bounds are derived for the8-PSK TCM system whose diversity is increased by usinga regular 2/3 rate convolutional code followed bythree bit interleavers and a 8-PSK signal mapperfollowed by two coordinate interleavers. Theanalytical and simulation results show that toincrease the diversity of the trellis coded M-PSKsystems in Rayleigh fading channels is the primary keyfor reliable transmission of high quality voice anddigital data.     

Analysis of the interaction between linear cyclic error correcting codes and self-synchronous payload scramblers

In order to protect public network data transmission from potential Layer 1 attacks by malicious users, selfsynchronous scramblers have come into widespread use. Such networks include those using ATM, Packet over SONET (POS), and the new Generic Framing Procedure (GFP). Unfortunately, feedback taps inherent in self-synchronous descramblers cause multiplication of transmission errors, which in turn degrades the performance of most linear cyclic error detection/correction codes. This paper analyzes this scrambler/code interaction with respect to the resulting probability of undetectable errors and transmission error correction capability. The theoretical criteria are derived for a linear cyclic code to maintain its error detection and correction performance in the presence of the scramblers. A novel approach for improving the error correction capabilities is also presented.     

Performance of an error correction scheme for PAL-TV teletext

The shortened (272, 190) difference set cyclic code has been adopted for the Japanese teletext system using NTSC vertical blanking periods. This system is known as BEST (burst and random error correction system for teletext) for NTSC-TV systems. A powerful error correction system adopting the (272, 190) code for PAL-TV teletext signals is described. To compare the error correction characteristics of this system to those of WST (World System Teletext), transmission experiments were carried out multiplexing both data packets with different lines during the vertical blanking period of the same TV signals. The validity of applying BEST to the PAL-TV signals was confirmed, particularly when there is ghost interference     

Burst error correction in high-speed digital mobile radio communications

In high-speed digital mobile radio communication, transmission performance is severely degraded by frequency-selective fading caused by the delay time spread of multipath propagation. Forward error correction (FEC) is one of the most effective techniques to improve transmission quality. The authors evaluate the effect of FEC on burst errors under frequency-selective fading. The FEC effect with diversity reception and interleaving is also investigated.<>     

A Time Compression Multiplexing System for a Circuit Switched Digital Capability

The time compression multiplexing (TCM) technique offers a solution to the problem of achieving digital penetration of the two-wire loop plant. TCM is planned to provide the digital loop transmission in the new circuit switched digital capability of the Bell System, so that a full-duplex 56 khit/s digital connection is obtained between subscriber's premises and the serving central office. Herein, the concept of TCM and its parameters are described. Tradeoffs leading to determination of system parameters such as burst frequency, burst length, guard time, and delay as they affect the range of the system are discussed. Also, the performance of the digital loops in terms of eye diagrams, system range limitations due to crosstalk and impulse noise, and percentage of loops covered are summarized. In addition, the circuit implementation using both analog and digital technologies is described. Experiences with various digital architectures to implement TCM and tradeoffs leading to a low-power digital architecture suitable for VLSI are discussed.     

Analysis of recurrent codes

A definition of a recurrent code is given in a framework which renders it amenable to mathematical analysis. Recurrent codes for both independent and burst errors are considered, and a necessary and sufficient condition for either type of error correction is established. For burst-error-correcting codes, the problem treated is (for a fixed burst length and redundancy) the minimization of the error-free distance ("guard space") required between bursts. A lower bound is obtained on the guard space, and in certain cases, codes which realize this bound are given. A general code which is close to the lower bound in many cases is also given. For independent errors, a code which will correct any error, provided that no consecutive "n" positions have more than "e" digits in error, is discussed. Fore = 1, a necessary and sufficient condition onnis derived; fore > 1, a lower bound onnis obtained, and for the case of redundancy1/2, an upper bound onnis also derived.     

Performance of Forward-Error Correction Code in 10-Gb/s RSOA-Based WDM PON

We investigate the performance of the forward-error correction (FEC) code for the 10-Gb/s wavelength-division- multiplexed passive optical network (WDM PON) implemented by using reflective semiconductor optical amplifiers (RSOAs) with extremely limited modulation bandwidth and the electronic equalizers to compensate for the degradations resulting from the use of such RSOAs. We show that the error occurrences in this network strongly depend on the bit pattern and the burst errors are likely to occur. Thus, it is important to use the FEC code capable of correcting the burst errors such as Reed–Solomon (RS) code. In addition, since a significant penalty can be induced by the increased line rate resulting from the use of the FEC code, it is necessary to find the optimum redundancy required to minimize the bit-error rate. We also evaluate the tolerance to the chromatic dispersion of the proposed 10-Gb/s WDM PON implemented by using the RS code with the optimum redundancy.      

Error statistics of 64-QAM signal in AM/64-QAM hybrid optical transmission

This paper describes the performance of an error correction system based on the DAVIC specifications. It also presents an error statistics property and its analytical model for the impulse noise induced by clipping in amplitude modulation (AM)/64 quadrature amplitude modulation (QAM) hybrid optical transmission (hybrid transmission). We found in hybrid transmission that the errors cannot be completely corrected by the error correction of DAVIC after hit error rate (BER) measurement with error correction, and then the error statistics of the 64-QAM channel were evaluated. It was confirmed that errors occasionally occur in long bursts that exceed an interleaver block length of the error correction when the impulse noise degrades the 64-QAM signal's BER. We believe that the error burst is caused by the slowed fluctuation of the envelope of a frequency division multiplex (FDM) signal in hybrid transmission. Once a portion of the envelope amplitude reaches the clipping level, the duration of maintaining such a condition is much longer than the 64-QAM symbol time. Accordingly, the errors occasionally run for more than 100 symbols. We apply Rice's fading duration distribution model to explain this error burst occurrence mechanism and its statistical manner and discuss this model's validity by comparison between the theoretical and measurement data of error statistics property. Consequently, we found that the theoretical value by the proposed model is in excellent agreement with the measurement data and confirmed that the proposed model is valid for analyzing the error manner for the clipping induced error.     

Digital frequency estimation in burst mode QPSK transmission

In burst digital transmission using PSK (phase shift keying) modulation with coherent detection, the recovery of the carrier reference phase and the symbol clock is a key aspect. If all users have a common clock synchronization, symbol timing needs not to be recovered in each burst. A digital processor for carrier recovery without preambles, in the presence of frequency offset, is considered. As an example, a 2 Mb/s QPSK transmission system is considered in which E _b/N_o=10 dB, and the burst and estimation interval length L=15. Using the algorithm described and averaging eight successive estimated frequency offsets, in order to eliminate anomalous errors, the BER (bit error rate) degradation is equal to 0.14 dB when Δf=20 kHz     

A CMOS switched-capacitor variable line equalizer

The authors describe the design concept and experimental results for a CMOS switched-capacitor variable-line equalizer to be used in time-compression multiplexed (TCM) digital subscriber loop transmission systems. The equalizer transfer function is optimized in the time domain to relax the filter complexity to half that required by the application of classical communication techniques. In order the equalize wide-bandwidth high-speed digital data, a 50 MHz CMOS operational amplifier is proposed. The amplifier uses a folded cascade and buffer structure to achieve good stability against load capacitance change. An experimental chip has been fabricated with 2.5 /spl mu/m CMOS technology. The chip shows excellent characteristics for the equalization of 200 kb/s data travelling through pair cables of 5 km and 0.4 mm diameter.     

Pseudo-analog speech transmission in mobile radio communicationsystems

A low-complexity pseudo-analog speech transmission scheme is proposed for portable communications. It uses a speech coder based on adaptive differential pulse code modulation (ADPCM) in combination with a multilevel digital modulation technique such as M-ary DPSK or M-ary FSK and features low quantization noise, bandwidth efficiency, and robustness to transmission errors. A nonsymmetric M -ary DPSK scheme called skewed M-ary DPSK is proposed to enhance the noisy channel performance. Comparison to conventional analog FM and a digital speech transmission scheme using adaptive predictive coding and forward error correction (FEC) based on convolutional coding shows that the pseudo-analog system has the best objective signal-to-noise ratio performance under most channel conditions. Informal subjective evaluations rate the digital system superior to the pseudo-analog scheme for bad channels and conversely for good channels. It is concluded that the pseudo-analog system can be designed with low delay and high speech quality for good channels with high spectral efficiency