Quality of service and bandwidth efficiency of cellular mobile radio with variable bit-rate speech transmission

The performance of an adjustable source/channel codec in a cellular mobile-radio environment is investigated. The speech transmission rate and the amount of forward error correction change in response to changing channel conditions. The channel rate is constant at 32 kb/s, and when the channel is good all of these bits are used for speech transmission. In intermediate and poor channels the speech rate is 24 or 16 kb/s, and the remaining channel symbols are used for forward error correction. Relative to conventional transmission this approach offers an improved grade of service. For example, the outage rate (the proportion of "poor or worse" communications) goes from nine percent with fixed-rate to three percent with variable-rate transmission. Alternatively, this improved grade of service can be exchanged for higher bandwidth efficiency. The fixed-rate system (with nine percent outage) has 23 users per cell. With 52 users per cell the outage of the variable-rate system is only six percent.     

The capacity of downlink fading channels with variable rate andpower

We obtain the Shannon capacity region of the down-link (broadcast) channel in fading and additive white Gaussian noise (AWGN) for time-division, frequency-division, and code-division. For all of these techniques, the maximum capacity is achieved when the transmitter varies the data rate sent to each user as their channels vary. This optimal scheme requires channel estimates at the transmitter; dynamic allocation of timeslots, bandwidth, or codes; and variable-rate and power transmission. For both AWGN and fading channels, nonorthogonal code-division with successive decoding has the largest capacity region, while time-division, frequency-division, and orthogonal code-division have the same smaller region. However, when all users have the same average received power, the capacity region for all these techniques is the same. In addition, the optimal nonorthogonal code is a multiresolution code which does not increase the signal bandwidth. Spread-spectrum code-division with successive interference cancellation has a similar rate region as this optimal technique, however, the region is reduced due to bandwidth expansion. We also examine the capacity region of nonorthogonal code-division without interference cancellation and of orthogonal code-division when multipath corrupts the code orthogonality. Our results can be used to bound the spectral efficiency of the downlink channel using time-division, frequency-division, and code-division, both with and without multiuser detection     

Coding for a Multiple-Access Frequency-Hopping System

A multiuser system employing on-off FSK modulation in conjunction with frequency hopping is examined for digital transmission. The receiver consists of a bank of bandpass noncoherent detectors followed by a message decoder. An analysis is presented for a transmission channel characterized by Rayleigh fading and additive Gaussian noise. The main source of impairment is interference between users. An upper limit on the number of simultaneous system users for acceptable transmission quality is derived. An example of a system with (one-way) bandwidth of 20 MHz and transmission rate of 32 kbits/s per user shows a maximum of 169 users at an average SNR on the channel of 25 dB and a bit error probability not exceeding 10^-3. The effect of applying error correcting codes to the system is evaluated. It is shown how Reed-Solomon codes can be used to generate both user identification and message coding. Such coding increases the number of users the system can accommodate in the example above to 212. Convolutional codes are shown to be even more effective. Such a code of constraint length 2 results in 285 simultaneous users in the example, which is an increase of 70 percent over an uncoded system. The drawback of coding is an increased complexity of the receiver. The amount of computation needed for the decoding of block and convolutional codes is estimated.     

Progressive transmission of images over memoryless noisy channels

An embedded source code allows the decoder to reconstruct the source progressively from the prefixes of a single bit stream. It is desirable to design joint source-channel coding schemes which retain the capability of progressive reconstruction in the presence of channel noise or packet loss. Here, we address the problem of joint source-channel coding of images for progressive transmission over memoryless bit error or packet erasure channels. We develop a framework for encoding based on embedded source codes and embedded error correcting and error detecting channel codes. For a target transmission rate, we provide solutions and an algorithm for the design of optimal unequal error/erasure protection. Three performance measures are considered: the average distortion, the average peak signal-to-noise ratio, and the average useful source coding rate. Under the assumption of rate compatibility of the underlying channel codes, we provide necessary conditions for progressive transmission of joint source-channel codes. We also show that the unequal error/erasure protection policies that maximize the average useful source coding rate allow progressive transmission with optimal unequal protection at a number of intermediate rates     

Embedded DPCM for Variable Bit Rate Transmission

Within the bit stream of an embedded digital code is a stream that can be decoded to produce a reasonable replica of the analog source signal. Unlike pulse code modulation (PCM), differential PCM (DPCM), is not an embedded code. IfCbits/sample are delected from the bit stream of a DPCM encoder withEbits/sample, the decoded analog signal is substantially noisier than the output of a DPCM codec withD = E - Cbits/sample. The penalty is 4-10 dB in signal-to-noise ratio (snr). However, with minor modifications to the encoder and decoder, DPCM becomes an embedded code. Embedded DPCM withEbits/ sample at the encoder andDbits/sample transmitted produces exactly the same output as embedded DPCM withDbits/sample encoding and perfect transmission. The snr of embedded DPCM is slightly lower than the Snr of DPCM. The penalty is 0.5-0.8 dB if the minimum transmitted bit rate is 2 bits/sample. It is less than 0.3 dB ifDis at least 3 bits/sample. Combined with an appropriate adaptive quantizer the embedded DPCM codec produces embedded ADPCM (adaptive DPCM) for variable rate transmission ranging from 2 bits/sample up to any desired maximum. Applications exist in speech interpolation, packet switching, and hardware architecture.     

Optimal use of Markov models for DPCM picture transmission over noisy channels

Joint source/channel decoders that use the residual redundancy in the source are investigated for differential pulse code modulation (DPCM) picture transmission over a binary symmetric channel. Markov sequence decoders employing the Viterbi algorithm that use first-order source statistics are reviewed, and generalized for decoders that use second-order source statistics. To make optimal use of the source correlation in both horizontal and vertical directions, it is necessary to generalize the conventional Viterbi decoding algorithm for a one higher-dimensional trellis. The paths through the trellis become two-dimensional "sheets", thus, the technique is coined "sheet decoding". By objective [reconstruction signal-to-noise ratio (SNR)] and subjective measure, it is found that the sheet decoders outperform the Markov sequence decoders that use a first-order Markov model, and outperform two other known decoders (modified maximum a posteriori probability and maximal SNR) that use a second-order Markov model. Moreover, it is found that the use of a simple rate-2/3 block code in conjunction with Markov model-aided decoding (MMAD) offers significant performance improvement for a 2-bit DPCM system. For the example Lenna image, it is observed that the rate-2/3 block code is superior to a rate-2/3 convolutional code for channel-error rates higher than 0.035. The block code is easily incorporated into any of the MMAD DPCM systems and results in a 2-bit MMAD DPCM system that significantly outperforms the noncoded 3-bit MMAD DPCM systems for channel-error rates higher than 0.04.     

Partial response continuous phase modulation and DPCM coding for speech transmission in cellular mobile radio systems

The performance of a speech transmission scheme with application to cellular digital mobile radio systems is considered. The source coder is embedded differential pulse code modulation (DPCM) and the modulation schemes belong to the class of partial response continuous phase modulation (CPM). Both quantizing noise and transmission errors contribute to the overall mean square error. The performance measure is the audio signal-to-noise ratio (SNR). It is seen that in a fading environment space diversity is very effective in bringing down the threshold of channel SNR to maintain the required audio SNR. The number of channels the system can support is evaluated under various conditions.     

Progressive source coding for a power constrained Gaussian channel

We consider the progressive transmission of a lossy source across a power constrained Gaussian channel using binary phase-shift keying modulation. Under the theoretical assumptions of infinite bandwidth, arbitrarily complex channel coding, and lossless transmission, we derive the optimal channel code rate and the optimal energy allocation per transmitted bit. Under the practical assumptions of a low complexity class of algebraic channel codes and progressive image coding, we numerically optimize the choice of channel code rate and the energy per bit allocation. This model provides an additional degree of freedom with respect to previously proposed schemes, and can achieve a higher performance for sources such as images. It also allows one to control bandwidth expansion or reduction     

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.     

Rate 3/4 Convolutional Coding of 16-PSK: Code Design and Performance Study

Convolutional coding coupled with 16-PSK modulation is investigated for bandwidth efficient transmission. Rate 3/4, small memory codes are found which are optimized in the free-distance sense on the Gaussian channel. These codes provide up to 4.8 dB of coding gain with 32 states over uncoded 8-PSK, a scheme having the same spectral efficiency as the codes described. The performance is compared with earlier findings of Ungerboeck and some recent results onR = 2/3coded 8-PSK. In addition, we present results of a channel transmission study to assess the performance of the four-state code on the band-limited nonlinear channel, and find that performance of the coded scheme degrades comparably with uncoded 8-PSK, i.e., coding gain is roughly preserved.     

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     

Combined Source-Channel Coding of Images

A combined source-channel coding approach is described for the encoding, transmission and remote reconstruction of image data. The source encoder employs two-dimensional (2-D) differential pulse code modulation (DPCM). This is a relatively efficient encoding scheme in the absence of channel errors. In the presence of channel errors, however, the performance degrades rapidly. By providing error control protection to those encoded bits which contribute most significantly to image reconstruction, it is possible to minimize this degradation without sacrificing transmission bandwidth. The result is a relatively robust design which is reasonably insensitive to channel errors and yet provides performance approaching the rate-distortion bound. Analytical results are provided for assumed 2-D autoregressive image models while simulation results are described for real-world images.     

Curves on a sphere, shift-map dynamics, and error control for continuous alphabet sources

We consider two codes based on dynamical systems, for transmitting information from a continuous alphabet, discrete-time source over a Gaussian channel. The first code, a homogeneous spherical code, is generated by the linear dynamical system s/spl dot/=As, with A a square skew-symmetric matrix. The second code is generated by the shift map s/sub n/=b/sub n/s/sub n-1/(mod 1). The performance of each of these codes is determined by the geometry of its locus or signal set, specifically, its arc length and minimum distance, suitably defined. We show that the performance analyses for these systems are closely related, and derive exact expressions and bounds for relevant geometric parameters. We also observe that the lattice /spl Zopf//sup N/ underlies both modulation systems and we develop a fast decoding algorithm that relies on this observation. Analytic results show that for fixed bandwidth expansion, good scaling behavior of the mean squared error is obtained relative to the channel signal-to-noise ratio (SNR). Particularly interesting is the resulting observation that sampled, exponentially chirped modulation codes are good bandwidth expansion codes.     

A vector quantization approach to universal noiseless coding andquantization

A two-stage code is a block code in which each block of data is coded in two stages: the first stage codes the identity of a block code among a collection of codes, and the second stage codes the data using the identified code. The collection of codes may be noiseless codes, fixed-rate quantizers, or variable-rate quantizers. We take a vector quantization approach to two-stage coding, in which the first stage code can be regarded as a vector quantizer that “quantizes” the input data of length n to one of a fixed collection of block codes. We apply the generalized Lloyd algorithm to the first-stage quantizer, using induced measures of rate and distortion, to design locally optimal two-stage codes. On a source of medical images, two-stage variable-rate vector quantizers designed in this way outperform standard (one-stage) fixed-rate vector quantizers by over 9 dB. The tail of the operational distortion-rate function of the first-stage quantizer determines the optimal rate of convergence of the redundancy of a universal sequence of two-stage codes. We show that there exist two-stage universal noiseless codes, fixed-rate quantizers, and variable-rate quantizers whose per-letter rate and distortion redundancies converge to zero as (k/2)n ^-1 log n, when the universe of sources has finite dimension k. This extends the achievability part of Rissanen's theorem from universal noiseless codes to universal quantizers. Further, we show that the redundancies converge as O(n^-1) when the universe of sources is countable, and as O(n^-1+ϵ) when the universe of sources is infinite-dimensional, under appropriate conditions     

Optimum rate Reed-Solomon codes for frequency-hoppedspread-spectrum multiple-access communication systems

The authors consider a multiple-access frequency-hopped spread-spectrum communication system with Reed-Solomon codes. The performance measures of interest are an achievable region and the channel throughput. The achievable rate region is the set of all pairs of code rate and number of users for which communication is possible with error probability below a fixed value. The throughput measures the expected number of successful codeword transmissions per unit bandwidth. Two models of interference are considered. For these two models, the authors determine the optimal number of users for a given bandwidth and the optimal rate Reed-Solomon code that maximize the throughput. They also determine the achievable region for these models     

Optimal resource allocation for wireless video over CDMA networks

We present a multiple-channel video transmission scheme in wireless CDMA networks over multipath fading channels. We map an embedded video bitstream, which is encoded into multiple independently decodable layers by 3D-ESCOT video coding technique, to multiple CDMA channels. One video source layer is transmitted over one CDMA channel. Each video source layer is protected by a product channel code structure. A product channel code is obtained by the combination of a row code based on rate compatible punctured convolutional code (RCPC) with cyclic redundancy check (CRC) error detection and a source-channel column code, i.e., systematic rate-compatible Reed-Solomon (RS) style erasure code. For a given budget on the available bandwidth and total transmit power, the transmitter determines the optimal power allocations and the optimal transmission rates among multiple CDMA channels, as well as the optimal product channel code rate allocation, i.e., the optimal unequal Reed-Solomon code source/parity rate allocations and the optimal RCPC rate protection for each channel. In formulating such an optimization problem, we make use of results on the large-system CDMA performance for various multiuser receivers in multipath fading channels. The channel is modeled as the concatenation of wireless BER channel and a wireline packet erasure channel with a fixed packet loss probability. By solving the optimization problem, we obtain the optimal power level allocation and the optimal transmission rate allocation over multiple CDMA channels. For each CDMA channel, we also employ a fast joint source-channel coding algorithm to obtain the optimal product channel code structure. Simulation results show that the proposed framework allows the video quality to degrade gracefully as the fading worsens or the bandwidth decreases, and it offers improved video quality at the receiver.     

Robust adaptive transmission of images and video over multiple channels

Reliable transmission of images and video over wireless networks must address both potentially limited bandwidths and the possibilities of loss. When bandwidth sufficient to transmit the bit stream is unavailable on a single channel, the data can be partitioned over multiple channels with possibly unequal bandwidths and error characteristics at the expense of more complex channel coding (i.e., error correction). This paper addresses the problem of efficiently channel coding and partitioning pre-encoded image and video bit streams into substreams for transmission over multiple channels with unequal and time-varying characteristics. Within channels, error protection is unequally applied based on both data decoding priority and channel packet loss rates, while cross-channel coding addresses channel failures. In comparison with conventional product codes, the resulting product code does not restrict the total encoded data to a rectangular structure; rather, the data in each channel is adaptively coded according to the channel's varying conditions. The coding and partitioning are optimized to achieve two performance criteria: maximum bandwidth efficiency and minimum delay. Simulation results demonstrate that this approach is effective under a variety of channel conditions and for a broad range of source material.     

Joint source and channel coding using turbo codes over rings

Turbo codes are a practical solution for achieving large coding gains. We present a new turbo coding scheme where the component codes are convolutional codes (CCs) over the ring of integers modulo M, with M being the alphabet size of the source encoder. The a priori knowledge of the source statistics is used during the iterative decoding procedure for improved decoder performance. As an example of application, we examine differential pulse code modulation (DPCM) encoded image transmission     

Two-Dimensional DPCM Image Transmission Over Fading Channels

A combined source-channel coding approach is described for the encoding, transmission, and remote reconstruction of image data. The transmission medium considered is that of a fading dispersive communications channel. Both the Rician fading and Rayleigh fading channel models are considered. The image source encoder employs two-dimensional (2-D) differential pulse code modulation (DPCM). This is a relatively efficient encoding scheme in the absence of channel errors. In the presence of fading, however, the performance degrades rapidly. By providing error control protection to those encoded bits which contribute most significantly to image reconstruction, it is possible to minimize this degradation without sacrificing transmission bandwidth. Several modulation techniques are employed in evaluation of system performance including noncoherent multiple frequency shift-keyed (MFSK) modulation. Analytical results are provided for assumed 2-D autoregressive image models, while simulation results are described for real-world images.