The Effect of Channel Coding on the Efficiency of Cellular Mobile Radio Systems

Error correcting codes have two opposite effects on the efficiency of cellular mobile-radio systems. Although they increase the bandwidth per channel, the codes also make signals more robust and thereby reduce the required distance between users of the same frequency band. This paper reports a mathematical study of the interactions of these two effects in determining the efficiencies of a large number of sourcecode and channel-code combinations. Within a statistical performance objective (basebandSNR geq 11dB for 90 percent of the users), the most efficient scheme in this study has an embedded differential pulse code modulation (DPCM) source code and a rate 1/2 channel code that protects 2 bits of each 4 bit DPCM code word. Based on a conservative model of cellular transmission, we estimate that the bandwidth efficiency is 3 users/cell/MHz of system bandwidth. By contrast, there are only 1.2 users/cell/MHz with uncoded transmission and 4.5 users/cell/MHz with a rather complicated variable-rate scheme. We also observe that the embedded source code, originally devised for variable-rate operation, has a higher baseband SNR than conventional DPCM in certain fixed-rate environments.     

Subband speech coding and matched convolutional channel coding formobile radio channels

The effects of digital transmission errors on a family of variable-rate embedded subband speech coders (SBC) are analyzed in detail. It is shown that there is a difference in error sensitivity of four orders of magnitude between the most and the least sensitive bits of the speech coder. As a result, a family of rate-compatible punctured convolutional codes with flexible unequal error protection capabilities have been matched to the speech coder. These codes are optimally decoded with the Viterbi algorithm. Among the results, analysis and informal listening tests show that with a 4-level unequal error protection scheme transmission of 12 kb/s speech is possible with very little degradation in quality over a 16 kb/s channel with an average bit error rate (BER) of 2×10^-2 at a vehicle speed of 60 m.p.h. and with interleaving over two 16 ms speech frames     

Service outage based power and rate allocation for parallel fading channels

The service outage based allocation problem explores variable-rate transmission schemes and combines the concepts of ergodic capacity and outage capacity for fading channels. A service outage occurs when the transmission rate is below a given basic rate r/sub o/. The allocation problem is to maximize the expected rate subject to the average power constraint and the constraint that the outage probability is less than /spl epsi/. A general class of probabilistic power allocation schemes is considered for an M-parallel fading channel model. The optimum power allocation scheme is derived and shown to be deterministic except at channel states of a boundary set. The resulting service outage achievable rate ranges from 1-/spl epsi/ of the outage capacity up to the ergodic capacity with increasing average power. Two near-optimum schemes are also derived by exploiting the fact that the outage probability is usually small. The second near-optimum scheme significantly reduces the computational complexity of the optimum solution; moreover, it has a simple structure for the implementation of transmission of mixed real-time and non-real-time services.     

Spread-Spectrum Mobile Radio with Variable-Bit-Rate Speech Transmission

We describe a digital mobile radio system in which users experiencing poor transmission reduce the number of bits per second used for speech coding from 32000 to 28000. They do so in a manner that greatly reduces the binary error rate. The reduced error rate more than compensates for increased quantization noise. We predict two statistics of user satisfaction in a mobile radio service area: POW, the percentage of users experiencing poor or worse transmission, and GOB, the percentage good or better. In a typical environment the maximum number of users that can operate within a 20 MHz band with POW ≤ 5 percent is 137, compared to 91 users when everyone transmits 32 kbits/s. The modulation technique is FH-FSK (frequency hopped, frequency shift keying) and the digital code is embedded ADPCM (adaptive differential pulse code modulation), in the mobile radio service area, signal strength depends on distance from mobile unit to a central base station. There is slow (shadow) fading due to terrain and building variations and rapid (Rayleigh) fading due to multipath propagation. To predict user satisfaction we have combined three different analyses: 1) a theory of FH-FSK, 2) a theory of the signal-to-noise ratio statistics in mobile radio service areas, and 3) a model, based on computer simulations and subjective testing data, of the dependence of subjective speech quality on ADPCM transmission rate and binary error rate.     

An adaptive modulation scheme for simultaneous voice and datatransmission over fading channels

We propose a new adaptive modulation technique for simultaneous voice and data transmission over fading channels and study its performance. The proposed scheme takes advantage of the time-varying nature of fading to dynamically allocate the transmitted power between the inphase (I) and quadrature (Q) channels. It uses fixed-rate binary phase shift keying (BPSK) modulation on the Q channel for voice, and variable-rate M-ary amplitude modulation (M-AM) on the I channel for data. For favorable channel conditions, most of the power is allocated to high rate data transmission on the I channel. The remaining power is used to support the variable-power voice transmission on the Q channel. As the channel degrades, the modulation gradually reduces its data throughput and reallocates most of its available power to ensure a continuous and satisfactory voice transmission. The scheme is intended to provide a high average spectral efficiency for data communications while meeting the stringent delay requirements imposed by voice. We present closed-form expressions as well as numerical and simulation results for the outage probability, average allocated power, achievable spectral efficiency, and average bit error rate (BER) for both voice and data transmission over Nakagami-m fading channels. We also discuss the features and advantages of the proposed scheme. For example, in Rayleigh fading with an average signal-to-noise ratio (SNR) of 20 dB, our scheme is able to transmit about 2 bits/s/Hz of data at an average BER of 10 ^-5 while sending about 1 bit/s/Hz of voice at an average BER of 10^-2     

流星突发通信中的变速率编码方案研究

流星突发通信是一种利用流星余迹进行通信的短时、突发通信方式。如何在较短通信时间内尽可能多的实现数据的可靠传输,始终是研究者关注的问题。传统的固定速率编码方案无法很好地适应信道变化,因此提出变速率编码以提高系统性能。通过对两种方案研究,对变速率编码算法进行了改进。仿真结果表明,改进后的变速率编码方案可明显改善系统性能。     

Variable-rate hybrid ARQ for meteor-burst communications

The authors introduce a form of automatic repeat request (ARQ), referred to as variable-rate type-I hybrid ARQ, in which the code rate varies in response to the fluctuations in the power received from a meteor trail. For one implementation, the source or the destination periodically obtains estimates of the signal power at the destination, and the source uses these estimates to select the rate of the code. For an alternative approach, the code rate is determined completely by the decoding successes and failures during previous transmissions. The performance measure is the throughput per trail, which is defined as the expected number of information bits received correctly for a given meteor trail. Numerical results for Reed-Solomon codes are included to illustrate the relative performance of the various schemes. It is shown that the throughput is larger for both implementations of variable-rate type-I hybrid ARQ than for fixed-rate type-I hybrid ARQ and ARQ without forward error correction     

Combined Speech and Channel Coding for Wireless Communications

In this paper, we propose acombined variable-rate code-excited linearly predictive (QCELP)speech coding and unequal error protection (UEP) channel codingsystem for wireless communications. In contrast to theconventional schemes, our system employs a concatenatedsuper-imposed rate-compatible punctured convolutional (SI-RCPC)channel coding scheme which can provide UEP with respect to notonly the bit-significance of speech packets but also the speechactivity and local channel characteristics. Verified by thesimulation results, the combined system achieves an averagetransmission rate less than 8 kb/s as well as an average 2 dBsignal-to-noise ratio (SNR) gain over the conventional equal errorprotection system.     

Huffman code based error screening and channel code optimization for error concealment in perceptual audio coding (PAC) algorithms

The class of perceptual audio coding (PAC) algorithms yields efficient and high-quality stereo digital audio bitstreams at bit rates from 16 kb/sec to 128 kb/sec (and higher). To avoid "pops and clicks" in the decoded audio signals, channel error detection combined with source error concealment, or source error mitigation, techniques are preferred to pure channel error correction. One method of channel error detection is to use a high-rate block code, for example, a cyclic redundancy check (CRC) code. Several joint source-channel coding issues arise in this framework because PAC contains a fixed-to-variable source coding component in the form of Huffman codes, so that the output audio packets are of varying length. We explore two such issues. First, we develop methods for screening for undetected channel errors in the audio decoder by looking for inconsistencies between the number of bits decoded by the Huffman decoder and the number of bits in the packet as specified by control information in the bitstream. We evaluate this scheme by means of simulations of Bernoulli sources and real audio data encoded by PAC. Considerable reduction in undetected errors is obtained. Second, we consider several configurations for the channel error detection codes, in particular CRC codes. The preferred set of formats employs variable-block length, variable-rate outer codes matched to the individual audio packets, with one or more codewords used per audio packet. To maintain a constant bit rate into the channel, PAC and CRC encoding must be performed jointly, e.g., by incorporating the CRC into the bit allocation loop in the audio coder.     

On the synergy between adaptive physical layer and multiple-access control for integrated voice and data services in a cellular wireless network

We propose a novel design to exploit the synergy between the multiple-access control (MAC) layer and the physical layer of a cellular wireless system with integrated voice and data services. As in a traditional design, the physical layer (channel encoder and modulator) is responsible for providing error protection for transmitting the packets over the hostile radio channel, while the MAC layer is responsible for allocating the precious bandwidth to the contending users for voice or data connections. However, a distinctive feature of our proposed design is that in the physical layer, a variable-rate adaptive channel encoder is employed to dynamically adjust the amount of forward error correction according to the time-varying wireless channel state such that the MAC layer, which is a reservation-based time-division multiple-access protocol, is able to make informed decisions as to bandwidth allocation. Specifically, based on the channel state information provided by the physical layer, the MAC protocol gives higher priority to users with better channel states. This novel synergistic mechanism between the two protocol layers can utilize the system bandwidth more effectively. The multiple-access performance of the proposed scheme is compared with two baseline systems. The first baseline system consists of the same reservation-based MAC protocol but with a traditional fixed-rate physical layer. The second system consists of the same reservation-based MAC protocol and the same channel adaptive physical layer, but without interaction between the two layers. All three protocols have a request queue, which stores the previous requests that survive the contention but are not allocated information slots. Our extensive simulation results demonstrate that significant performance gains are achieved through the exploitation of the synergy between the two protocol layers.     

Variable-rate coding for meteor-burst communications

The performance of variable-rate Reed-Solomon error-control coding for meteor-burst communications is considered. The code rate is allowed to vary from codeword to codeword within each packet, and the optimum number of codewords per packet and optimum rates for the codewords are determined as a function of the length of the message and the decay rate for the meteor trail. The resulting performance is compared to that obtained from, fixed-rate coding. Of central importance is the derivation of tractable expressions for the probability of correct decoding for bounded-distance decoding on a memoryless channel with a time-varying symbol error probability. A throughout measure is developed that is based on the probability distribution of the initial signal-to-noise ratio     

Optimal power allocation in DS-CDMA with adaptive SIC technique

This paper proposes an optimal power allocation in direct sequence-code division multiple access (DS-CDMA) system. The objective is to minimize total transmit power, while simultaneously meeting the certain sum channel capacity (data transmission rate) and outage probability constraints on Rayleigh fading channel. Then a weighted correlator with an adaptive successive interference cancelation (SIC) scheme is developed using neural network (NN) for an improvement in receiver performance. A closed mathematical form of joint probability of error (JPOE) is derived. This determines the number of active users’ interfering effect that needs to be canceled in order to achieve a desired bit error rate (BER) value. Mathematical analysis shows that better receiver performance can be achieved through large change in weight up-gradation (w) for the strong users with a particular change in learning rate (η). Simulation results in terms of sum capacity as well as weak user’s (users with poor channel gain) capacity, outage probability and BER performance duly support the effectiveness of the proposed scheme over the existing works.     

New results on coding of stationary nonergodic sources

Two results on the coding of stationary nonergodic sources are presented. The first is a source coding theorem stating that there exist variable-rate codes with performance arbitrarily close to the rate-distortion function of the stationary nonergodic source. The second is a converse information transmission theorem. It is shown that the distortion which results when the source is transmitted across a channel with capacityCis no less than the least distortion achievable by fixed-rate codes with rateC.     

Communication Over MIMO Broadcast Channels Using Lattice-Basis Reduction

A new viewpoint for adopting the lattice reduction in communication over multiple-input multiple-output (MIMO) broadcast channels is introduced. Lattice basis reduction helps us to reduce the average transmitted energy by modifying the region which includes the constellation points. The new viewpoint helps us to generalize the idea of lattice-reduction-aided (LRA) precoding for the case of unequal-rate transmission, and obtain analytic results for the asymptotic behavior (signal-to-noise ratio (SNR) $longrightarrow infty$) of the symbol error rate for the LRA precoding and the perturbation technique. Also, the outage probability for both cases of fixed-rate users and fixed sum rate is analyzed. It is shown that the LRA method, using the Lenstra–Lenstra–LovÁsz (LLL) algorithm, achieves the optimum asymptotic slope of symbol error rate (called the precoding diversity).      

Effects of PRMA on objective speech quality

When transmitting 32 kbit/s adaptive differential pulse code modulation (ADPCM) speech using Reed-Solomon error correction coding and 16 level quadrature amplitude modulation (16-QAM), a 20 slot packet reservation multiple access (PRMA) assisted cordless telecommunications (CT) scheme supported 36-38 speech users with negligible objective and subjective speech degradation. The average number of users per slot was nearly doubled due to deploying PRMA and toll quality speech was transmitted in a user bandwidth approximately 11.6 kHz. For a channel signal-to-noise ratio (SNR) in excess of 25 dB, a Rayleigh fading channel and mobile speeds above 2 mph the speech segmental SNR degradation was less than 0.3 dB.<>     

Variable-rate variable-power MQAM for fading channels

We propose a variable-rate and variable-power MQAM modulation scheme for high-speed data transmission over fading channels. We first review results for the Shannon capacity of fading channels with channel side information, where capacity is achieved using adaptive transmission techniques. We then derive the spectral efficiency of our proposed modulation. We show that there is a constant power gap between the spectral efficiency of our proposed technique and the channel capacity, and this gap is a simple function of the required bit-error rate (BER). In addition, using just five or six different signal constellations, we achieve within 1-2 dB of the maximum efficiency using unrestricted constellation sets. We compute the rate at which the transmitter needs to update its power and rate as a function of the channel Doppler frequency for these constellation sets. We also obtain the exact efficiency loss for smaller constellation sets, which may be required if the transmitter adaptation rate is constrained by hardware limitations. Our modulation scheme exhibits a 5-10-dB power gain relative to variable-power fixed-rate transmission, and up to 20 dB of gain relative to nonadaptive transmission. We also determine the effect of channel estimation error and delay on the BER performance of our adaptive scheme. We conclude with a discussion of coding techniques and the relationship between our proposed modulation and Shannon capacity     

一种提高协作通信性能的新方法

传统的无线通信中,由于外部环境的干扰,信道的不稳定,网络的中断率比较高。在信道条件较差时,通信性能较差。本文利用无线通信节点间的合作,通过不同节点对信息的中继放大、解码（或者网络编码）,对直接传输的无线信道性能进行改进,大大提高了网络性能,降低网络的中断率;通过比较不同频谱效率下的不同转发方式的中断率性能,选择合适的传输方法。仿真实验得到：频谱效率小于一定值的情况下,利用放大和网络编码转发方式的性能大大高于直接传输方式。     

Performance analysis of the adaptive EXP/PF channel scheduler in an AMC/TDM system

The adaptive EXP/PF AU; PL. DEFINE "EXP" ALSO channel scheduler proposed in Rhee et al. (2003) is an enhancement to the proportional fairness (PF) channel scheduler in that it guarantees delay sensitive services having the desired service delay outage probability as well as best-effort services in the forward link of an adaptive modulation and coding when used with time-division multiplexing (AMC/TDM) system. In this letter, assuming that there are many best-effort service users and one streaming service user requiring maximum delay constraints of 3 s, the system throughput of the adaptive EXP/PF scheduler is evaluated for different channel conditions of the streaming service user and is also compared with PF scheduler and its upper bound of throughput. The results show that the adaptive EXP/PF algorithm offers high system throughput as well as fairness among best-effort service users and guarantees streaming services of a desired delay constraint and outage probability.     

Image coding using variable-rate side-match finite-state vectorquantization

Future B-ISDN (broadband integrated services digital network) users will be able to send various kinds of information, such as voice, data, and image, over the same network and send information only when necessary. It has been recognized that variable-rate encoding techniques are more suitable than fixed-rate techniques for encoding images in a B-ISDN environment. A new variable-rate side-match finite-state vector quantization with a block classifier (CSMVQ) algorithm is described. In an ordinary fixed-rate SMVQ, the size of the state codebook is fixed. In the CSMVQ algorithm presented, the size of the state codebook is changed according to the characteristics of the current vector which can be predicted by a block classifier. In experiments, the improvement over SMVQ was up to 1.761 dB at a lower bit rate. Moreover, the improvement over VQ can be up to 3 dB at nearly the same bit rate.     

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