We address the problem of synthesis and generation of faces from edgemaps, motivated by extreme low bit-rate facial compression and the need for robust source-channel coding over noisy channels. Three approaches for image reconstruction are proposed. In the first, a deep learning-based encoder-decoder creates a latent space representation of the original image. An Edgemap-to-Latent Mapper (ELM) network maps the input edgemap to this latent space, with the final image reconstructed using a pre-trained compressive decoder. The second approach retrains the compressive decoder to reconstruct images from the ELM network’s output. The third approach jointly trains the ELM network and decoder, enabling direct reconstruction from the edgemap. This end-to-end framework achieves reasonable reconstruction fidelity. We also examine the impact of additive channel noise on edgemap transmission under low SNR conditions, demonstrating that even with significant noise, a DNN-based joint denoiser and edgemap decoder can reconstruct images. At extremely low SNRs, where edgemaps are highly corrupted, the network also exhibits generative capabilities, producing plausible images.

In the context of cellular networks, users located at the periphery of cells are particularly vulnerable to substantial interference from neighbouring cells, which can be represented as a two-user interference channel. This study introduces two highly effective methodologies, namely TwinNet and SiameseNet, using autoencoders, tailored for the design of encoders and decoders for block transmission and detection in interference-limited environments. The findings unambiguously illustrate that the developed models are capable of leveraging the interference structure to outperform traditional methods reliant on complete orthogonality. While it is recognized that systems employing coordinated transmissions and independent detection can offer greater capacity, the specific gains of data-driven models have not been thoroughly quantified or elucidated. This paper conducts an analysis to demonstrate the quantifiable advantages of such models in particular scenarios. Additionally, a comprehensive examination of the characteristics of codewords generated by these models is provided to offer a more intuitive comprehension of how these models achieve superior performance.

Orthogonal Time Frequency Space (OTFS) receivers perform joint equalization of all the symbols in an OTFS frame. We consider delay-time domain Maximal Ratio Combining (MRC) equalization for OTFS systems with receiver oversampling. The associated impairments are Inter-Frame Interference (IFI) and Inter-Block Interference (IBI). The existing Zero-Padding (ZP) approach avoids IFI and IBI but requires high overhead. We propose a Reduced Cyclic Prefix (RCP) and Reduced Cyclic Suffix (RCS) insertion method on the transmitter side that achieves high throughputs. Our approach avoids IFI, and we modify the existing equalizer designed for ZP systems to handle IBI without a significant increase in complexity. We present the Bit Error Rate (BER) simulation results for different oversampling factors and explain the diminishing returns.

The massive multiple-input multiple-output (MIMO) full-duplex two-way relaying (FD-TWR) literature has extensively investigated power scaling for rate guarantees by considering a fixed number of users. We investigate the pairwise error probability (PEP) and the per-user rate of a FD-TWR with N, relay antennas that employs maximal ratio combining/transmission to enable two-way communication between K FD users. We propose novel relay and user powers scalings, with both N, and K tending to infinity, and show that the PEP of each user converges almost surely to its AWGN counterpart. These power scalings are different from the existing ones, which are derived by fixing K and by assuming that only N, tends to large values. We show that the analysis developed herein applies to both Gaussian and non-Gaussian complex channels with finite number of moments. We numerically show that when both K and N, increase concurrently to large values, the proposed power scaling schemes not only have better per-user PEP and rate than the existing schemes, but they are also robust to the FD self loop-interference power.

Faster-Than-Nyquist (FTN) Signaling is a non-orthogonal transmission scheme which violates the Nyquist zero-ISI criterion providing higher throughput and better spectral efficiency than a Nyquist transmission scheme. This comes with a cost of higher transceiver complexity. In this paper, we focus on understanding pulse shapes and their inter-symbol-interference (ISI) and show that, under certain conditions on pulse shapes and τ (time acceleration factor), the ISI can be avoided completely with the help of precoding. This leads to a symbol-by-symbol detection. Further, we extend this idea to Orthogonal Frequency Division Multiplexing (OFDM) FTN systems and show that, under certain conditions, the average performance of OFDM system reaches that of a Nyquist system. Finally, simulation results of the performance of precoded and non-precoded single carrier and OFDM FTN systems are compared to a Nyquist system.

Theoretical analysis has long indicated that feedback improves the error exponent but not the capacity of memoryless Gaussian channels. Recently, Chen et al. demonstrated that an incremental redundancy scheme can use noiseless feedback to help short convolutional codes deliver the bit-error-rate performance of a long blocklength turbo code, but with much lower latency. Such a latency improvement is suggested by the error-exponent analysis, but there is no theoretical work that estimates how much latency improvement is possible with feedback for practical blocklengths and rates. This paper provides a code-independent analysis that quantifies the latency benefits possible by using modified incremental redundancy with feedback (MIRF). A sphere-packing analysis yields the throughput vs. latency performance of both a baseline ACK/NACK scheme and MIRF. The sphere-packing analysis matches well with simulations using turbo and convolutional codes, showing that the analysis has a practical predictive value.

Transmit diversity schemes with multiple streams obtain reliable transmission but suffer from interference when decoding multiple streams simultaneously. In this paper, we present closed-loop space-time block codes for multiple streams with limited channel information in order to ease suppressing interference at a receiving station. The proposed scheme includes constellation rotation in Alamouti coding pairs by feedback angle information. The sub-optimum solution for feedback angle is found to maximize the output signal-to-noise ratio (SNR). Simulation results show the proposed scheme with only a few bits feedback achieves nearly full diversity gain with reduced interference.

In this paper, new space time/frequency block coding (STBC/SFBC) scheme with angle feedback for 4-transmit antennas is proposed. This scheme achieves the full diversity gain with the full code rate by rotating constellation symbols in STBC/SFBC codewords using the quasi-orthogonal structure. With multiple receive antennas, this scheme can also be applied with additional receive antenna diversity. We show the channel characteristics can be much improved with as small as 1-bit feedback overhead. Simulation shows that the proposed scheme with 1-bit feedback achieves near-optimum performance.

Space-time coding is a bandwidth and power efficient method of communication over fading channels that realizes the benefits of multiple transmit antennas. Specific codes have been constructed using design criteria derived for quasi-static flat Rayleigh or Rician fading, where channel state information is available at the receiver. It is evident that the practicality of space-time codes will be greatly enhanced if the derived design criteria remain valid in the absence of perfect channel state information. It is even more desirable that the design criteria not be unduly sensitive to frequency selectivity and to the Doppler spread. This paper presents a theoretical study of these issues beginning with the effect of channel estimation error. Here it is assumed that a channel estimator extracts fade coefficients at the receiver and for constellations with constant energy, it is proved that in the absence of ideal channel state information the design criteria for space-time codes is still valid. The analysis also demonstrates that standard channel estimation techniques can be used in conjunction with space-time codes provided that the number of transmit antennas is small. We also derive the maximum-likelihood detection metric in the presence of channel estimation errors. Next, the effect of multiple paths on the performance of space-time codes is studied for a slowly changing Rayleigh channel. It is proved that the presence of multiple paths does not decrease the diversity order guaranteed by the design criteria used to construct the space-time codes. Similar results hold for rapid fading channels with or without multiple paths. The conclusion is that the diversity order promised by space-time coding is achieved under a variety of mobility conditions and environmental effects.

We introduce a new class of space-time codes called super-orthogonal space-time trellis codes. These codes combine set partitioning and a super set of orthogonal space-time block codes in a systematic way to provide full diversity and improved coding gain over earlier space-time trellis code constructions. We also study the optimality of our set partitioning and provide coding gain analysis. Codes operating at different rates, up to the highest theoretically possible rate, for different number of states can be designed by using our optimal set partitioning. Super-orthogonal space-time trellis codes can provide a tradeoff between rate and coding gain. Simulation results show more than 2-dB improvements over the codes presented in the literature while providing a systematic design methodology.

Real world source coding algorithms usually leave a certain amount of redundancy within the coded bit stream. Shannon (1948) already mentioned that this redundancy can be exploited at the receiver side to achieve a higher robustness against channel errors. We show how joint source-channel decoding can be performed in a way that is applicable to any mobile communication system standard. Considerable gains in terms of bit error rate or signal-to-noise ratio (SNR) are possible dependent on the amount of redundancy. However, an even better performance can be achieved by changing also the transmitter sided source and channel encoders. We propose an encoding concept employing low-dimensional quantization. Keeping the gross bit rate as well as the clean channel quality the same, it decreases the complexity of the source encoder and the decoder significantly. Finally, we give an application of our methods to spectral coefficient coding in speech transmission over a Rayleigh fading channel resulting in channel SNR gains of about 2 dB as compared to state-of-the-art (de-)coding and bad frame handling methods.

Abstract The complexity of channel equalization using the BCJR algorithm [Bahl LR et al. IEEE Transactions on Info. Theory 1974; 20: 284–287] grows exponentially with the channel memory. The M‐BCJR algorithm [Franz V, Anderson J. IEEE J. Selected Areas Comm. 1998; 16(2): 186–195] provides a method for reduced‐trellis channel equalization, but its performance varies significantly with the distribution of a channel energy to its taps. This paper proposes a variation of the M‐algorithm, based on the implementation of the BCJR algorithm in the logarithmic domain, which can offer robust performance. Designer choices such as the decision delay, maximum and minimum phase transformation, and selection of states are discussed. Simulation results demonstrate the effect of different parameters on the proposed algorithm's performance. Copyright © 2001 John Wiley & Sons, Ltd.

In digital transmission of speech, audio, images and video signals residual redundancy is often left after source coding due to the complexity and delay constraints. This redundancy remains both inside one block or frame but also in a time correlation of subsequent frames. We describe an approach to improve channel and source decoding by using both kinds of correlation. Further on we consider the bit-mapping and the multiplexing in coding and its effect on decoding. The mutual information as measurement for the gain in decoding through a priori information is explained. In this work on combined source and channel decoding, we try to answer the following question: can a priori information that models the source parameters be used twice; first at the channel decoder and then at the source decoder. The channel decoder uses the a priori information that models the bit stream generated by the source coder. This does not capture all the details of the source parameter level statistics. By exploiting the a priori knowledge of parameters (once more) at the source decoder, we show that it is possible to achieve better reconstruction than if this information was used at either of the decoders.

Wireless systems are subject to a time-varying and unknown a priori combination of cochannel interference, fading, and Gaussian noise. It is well known that multiple antennas can provide diversity in space that allows system tradeoffs between interference suppression and mitigation of fading. This paper describes how to achieve these same tradeoffs through diversity in time provided by channel coding. The mathematical description of time diversity is identical to that of space diversity, and what emerges is a unified framework for signal processing. Decoding algorithms are provided for repetition codes, rate 1/n convolutional codes, first-order Reed-Muller codes, and a new class of linear combination codes that provide cochannel interference suppression. In all cases it is possible to trade performance for complexity by choosing between joint estimation and a novel low-complexity linear canceler structure that treats interference as noise. This means that a single code can be used in a variety of system environments just by changing the processing in the receiver.

Transmitter diversity is an effective technique to improve wireless communication performance. In this paper, we investigate transmitter diversity using space-time coding for orthogonal frequency division multiplexing (OFDM) systems in high-speed wireless data applications. We develop channel parameter estimation approaches, which are crucial for the decoding of the space-time codes, and we derive the MSE bounds of the estimators. The overall receiver performance using such a transmitter diversity scheme is demonstrated by extensive computer simulations. For an OFDM system with two transmitter antennas and two receiver antennas with transmission efficiency as high as 1.475 bits/s/Hz, the required signal-to-noise ratio is only about 7 dB for a 1% bit error rate and 9 dB for a 10% word error rate assuming channels with two-ray, typical urban, and hilly terrain delay profiles, and a 40-Hz Doppler frequency. In summary, with the proposed channel estimator, combining OPDM with transmitter diversity using space-time coding is a promising technique for highly efficient data transmission over mobile wireless channels.

Space-time coding is a bandwidth and power efficient method of communication over fading channels that realizes the benefits of multiple transmit antennas. Specific codes have been constructed using design criteria derived for quasi-static flat Rayleigh or Rician fading, where channel state information is available at the receiver. It is evident that the practicality of space-time codes will be greatly enhanced if the derived design criteria remain valid in the absence of perfect channel state information. It is even more desirable that the design criteria not be unduly sensitive to frequency selectivity and to the Doppler spread. This paper presents a theoretical study of these issues beginning with the effect of channel estimation error. Here it is assumed that a channel estimator extracts fade coefficients at the receiver and for constellations with constant energy, it is proved that in the absence of ideal channel state information the design criteria for space-time codes is still valid. The analysis also demonstrates that standard channel estimation techniques can be used in conjunction with space-time codes provided that the number of transmit antennas is small. We also derive the maximum-likelihood detection metric in the presence of channel estimation errors. Next, the effect of multiple paths on the performance of space-time codes is studied for a slowly changing Rayleigh channel. It is proved that the presence of multiple paths does not decrease the diversity order guaranteed by the design criteria used to construct the space-time codes. Similar results hold for rapid fading channels with or without multiple paths. The conclusion is that the diversity order promised by space-time coding is achieved under a variety of mobility conditions and environmental effects.

The information capacity of wireless communication systems may be increased dramatically by employing multiple transmit and receive antennas. The goal of system design is to exploit this capacity in a practical way. An effective approach to increasing data rate over wireless channels is to employ space-time coding techniques appropriate to multiple transmit antennas. These space-time codes introduce temporal and spatial correlation into signals transmitted from different antennas, so as to provide diversity at the receiver, and coding gain over an uncoded system. For large number of transmit antennas and at high bandwidth efficiencies, the receiver may become too complex whenever correlation across transmit antennas is introduced. This paper dramatically reduces encoding and decoding complexity by partitioning antennas at the transmitter into small groups, and using individual space-time codes, called the component codes, to transmit information from each group of antennas. At the receiver, an individual space-time code is decoded by a novel linear processing technique that suppresses signals transmitted by other groups of antennas by treating them as interference. A simple receiver structure is derived that provides diversity and coding gain over uncoded systems. This combination of array processing at the receiver and coding techniques for multiple transmit antennas can provide reliable and very high data rate communication over narrowband wireless channels. A refinement of this basic structure gives rise to a multilayered space-time architecture that both generalizes and improves upon the layered space-time architecture proposed by Foschini (see Bell Labs Tech. J., vol.1, no.2, 1996).

This paper presents the theory and practice of a new advanced modem technology suitable for high-data-rate wireless communications and presents its performance over a frequency-flat Rayleigh fading channel. The new technology is based on space-time coded modulation (STCM) with multiple transmit and/or multiple receive antennas and orthogonal pilot sequence insertion (O-PSI). In this approach, data is encoded by a space-time (ST) channel encoder and the output of the encoder is split into N streams to be simultaneously transmitted using N transmit antennas. The transmitter inserts periodic orthogonal pilot sequences in each of the simultaneously transmitted bursts. The receiver uses those pilot sequences to estimate the fading channel. When combined with an appropriately designed interpolation filter, accurate channel state information (CSI) can be estimated for the decoding process. Simulation results of the proposed modem, as applied to the IS-136 cellular standard, are presented. We present the frame error rate (FER) performance results as a function of the signal-to-noise ratio (SNR) and the maximum Doppler frequency, in the presence of timing and frequency offset errors. Simulation results show that for a 10% FER, a 32-state eight-phase-shift keyed (8-PSK) ST code with two transmit and two receive antennas can support data rates up to 55.8 kb/s on a 30-kHz channel, at an SNR of 11.7 dB and a maximum Doppler frequency of 180 Hz. Simulation results for other codes and other channel conditions are also provided. We also compare the performance of the proposed STCM scheme with delay diversity schemes and conclude that STCM can provide significant SNR improvement over simple delay diversity.

We consider the design of channel codes for improving the data rate and/or the reliability of communications over fading channels using multiple transmit antennas. Data is encoded by a channel code and the encoded data is split into n streams that are simultaneously transmitted using n transmit antennas. The received signal at each receive antenna is a linear superposition of the n transmitted signals perturbed by noise. We derive performance criteria for designing such codes under the assumption that the fading is slow and frequency nonselective. Performance is shown to be determined by matrices constructed from pairs of distinct code sequences. The minimum rank among these matrices quantifies the diversity gain, while the minimum determinant of these matrices quantifies the coding gain. The results are then extended to fast fading channels. The design criteria are used to design trellis codes for high data rate wireless communication. The encoding/decoding complexity of these codes is comparable to trellis codes employed in practice over Gaussian channels. The codes constructed here provide the best tradeoff between data rate, diversity advantage, and trellis complexity. Simulation results are provided for 4 and 8 PSK signal sets with data rates of 2 and 3 bits/symbol, demonstrating excellent performance that is within 2-3 dB of the outage capacity for these channels using only 64 state encoders.

We revisit space-time codes for a mobile communication system that employs multiple antennas at the base station and optional antenna diversity at the mobile station. The realistic case when the channel state is not completely known is considered. It is assumed that the channel estimator extracts the fade coefficients using orthogonal pilot tones. Mismatch analysis is then carried out. It is proved that in the absence of ideal channel state information the design criteria for space-time codes developed in Tarokh et al. (1997) is still valid for the equal energy constellation case. Using our derivation, it is observed that channel estimation techniques commonly used over rapidly fading channels can be used in conjunction with space-time codes provided that the number of transmit antennas is small.

We consider the design of channel codes for improving the data rate and/or the reliability of communications over fading channels using multiple transmit antennas. Here, data is encoded by a channel code and the encoded data is split into n streams that are simultaneously transmitted using n transmit antennas. The received signal at each receive antenna is a linear superposition of the n transmitted signals. We derive performance criteria for designing channel codes under the assumption that the fading is slow and frequency non-selective. Performance is shown to be determined by diversity gain quantified by ranks and coding gain quantified by determinants of certain matrices that are constructed from the code sequences.

We consider the design of channel codes for improving the data rate and/or the reliability of communications using multiple transmit antennas over a fading channel. It is assumed that the transmitter does not know the channel but seeks to choose a codebook that guarantees a diversity gain of r/sub 1/ when there is no mobility and a diversity gain of r/sub 2//spl ges/r/sub 1/ when the channel is fast fading. A solution to this problem is unveiled in this paper. Here, the encoded data is split into n streams that are simultaneously transmitted using n transmit antennas. The signal received at each receive antenna is a superposition of the faded versions of the n transmitted signals. We derive performance criteria for designing codes having the aforementioned properties. Performance is shown to be determined by diversity advantage quantified by a rank/distance and coding advantage quantified by a determinant/product criterion. The criteria is used to design codes for both slow and rapid fading channels. The constructed codes have remarkable performance in low signal to noise ratios and are suitable for improving the frequency reuse factor under a variety of mobility conditions.

We develop a theory of the error rate of receivers that use tentative decisions for adaptive cancellation of intersymbol interference (ISI). We formulate precise conditions under which tentative decisions can be effectively used to cancel linear or nonlinear ISI, and verify the predictions of the theory by Monte Carlo simulations on four typical channels. In the case of magnetic recording channels, we show that the conditions for effective cancellation are satisfied by the precursor interference generated by inductive read heads, resulting in an improvement of the noise margin of 2 dB at the output of a symbol-by-symbol detector or 1 dB at the output of a Viterbi decoder at a bit error rate of 10/sup -9/. This technique has been incorporated in an experimental VLSI partial response maximum-likelihood (PRML) receiver, and the results confirmed by laboratory measurements. We also show that no improvement is obtained when this technique is used to compensate nonlinear distortion, which is another common impairment of magnetic recording channels, and we establish fundamental limits for the improvement achievable by any nonlinear equalization technique in magnetic recording.

An important problem in speech coding is the quantization of linear predictive coefficients (LPCs) with the smallest possible number of bits while maintaining robustness to a large variety of speech material and transmission media. Since direct quantization of LPCs is known to be unsatisfactory, the authors consider this problem for an equivalent representation, namely, the line spectral frequencies (LSFs). To achieve an acceptable level of distortion a scalar quantizer for LSFs requires a 36 bit codebook. The authors derive a 30 bit two-quantizer scheme which achieves a performance equivalent to this scalar quantizer. The two-quantizer format consists of both a vector and a scalar quantizer such that for each input, the better quantizer is used. The vector quantizer is designed from a training set that reflects the joint density (for coding efficiency) and which ensures coverage (for robustness). The scalar quantizer plays a pivotal role in dealing better with regions of the space that are sparsely covered by its vector quantizer counterpart. A further reduction of 1 bit is obtained by formulating a new adaptation algorithm for the vector quantizer and doing a dynamic programming search for both quantizers. The method of adaptation takes advantage of the ordering of the LSFs and imposes no overhead in memory requirements. The dynamic programming search is feasible due to the ordering property. Subjective tests in a speech coder reveal that the 29 bit scheme produces equivalent perceptual quality to that when the parameters are unquantized.>

Previously, we (see Wong, Sundberg and Seshadri, GLOBECOM '93, p.1649-1653, 1993) proposed a low-delay multiple access scheme, called shared time-division duplexing (STDD). This scheme allows both the up-link and downlink traffic to share a common channel, thereby achieving high statistical multiplexing gain even with a low population of simultaneous conversations. Here, we focus on transmitting speech using this multiple access scheme by a suitable choice of speech coder, channel coder and modulation. In particular, we consider low delay, high quality speech coding and digital modulation systems based on adaptive DPCM, with QDPSK or pseudo-analog transmission (skewed DPSK). 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%.>

A novel method of maximum likelihood sequence estimation is proposed for data that are transmitted over unknown linear channels. This procedure does not require a startup sequence for estimating the channel impulse response. Rather, the data and the channel are simultaneously estimated. It is implemented without any loss of optimality by a trellis search algorithm which searches for the best data sequence from among a number of hypothesized trellises which are constructed from the observed sequence. The number of states in each trellis and the number of trellises grow exponentially with the channel memory. A suboptimal trellis search algorithm is proposed whose complexity at best is slightly higher than that of the adaptive Viterbi algorithm operating with a known channel response. A simplified channel estimation algorithm when the number of data alphabets is greater than 2 is also proposed. Fast convergence of the algorithm in estimating the channel is demonstrated for binary pulse amplitude modulation over a variety of channels. Convergence over a wide range of SNR occurs within 100 symbols. The channel estimation algorithm for multi-level signals converges within 500-1000 symbols. We finally present an application of this algorithm to the problem of sequence estimation in the presence of rapidly time-varying intersymbol interference. The algorithm provides reliable zero-delay decisions which results in a better channel tracking algorithm when compared to previously proposed schemes.>

A list Viterbi decoding algorithm (LVA) produces a rank ordered list of the L globally best candidates after a trellis search. The authors present two such algorithms, (i) a parallel LVA that simultaneously produces the L best candidates and (ii) a serial LVA that iteratively produces the k/sup th/ best candidate based on knowledge of the previously found k-1 best paths. The application of LVA to a concatenated communication system consisting of an inner convolutional code and an outer error detecting code is considered in detail. Analysis as well as simulation results show that significant improvement in error performance is obtained when the inner decoder, which is conventionally based on the Viterbi algorithm (VA), is replaced by the LVA. An improvement of up to 3 dB is obtained for the additive white Gaussian noise (AWGN) channel due to an increase in the minimum Euclidean distance. Ever larger gains are obtained for the Rayleigh fading channel due to an increase in the time diversity. It is also shown that a 10% improvement in throughput is obtained along with significantly reduced probability of a decoding failure for a hybrid FEC/ARQ scheme with the inner code being a rate compatible punctured convolutional (RCPC) code.>

The authors present coded 8-phase-shift-keyed (8-PSK) modulations for the Rayleigh fading channel. The schemes are based on multilevel trellis-coded-modulation constructions and utilize maximum free Hamming distance binary convolutional codes as building blocks. A suboptimal multistage decoder that utilizes interstage interleaving and iterative decoding is proposed and evaluated. Examples are constructed to show that the proposed schemes outperform the best modified codes of the Ungerboeck type due to significantly higher implicit time diversity, yielding seven branches of built-in time diversity, whereas the Ungerboeck code yields four branches of time diversity for a 64-state system. The transmission delay is higher, however. The new schemes can provide three levels of unequal error protection when 8-PSK or 8-differential-phase-shift-keying (8-DPSK) modulations are used. They provide 10-14-dB channel signal-to-noise ratio gain over uncoded 4-DPSK at a bit error rate of 10/sup -3/ for a modest decoding complexity.>

Two combined unequal error protection (UEP) coding and modulation schemes are proposed. The first method multiplexes different coded signal constellations, with each coded constellation providing a different level of error protection. In this method, a codeword specifies the multiplexing rule and the choice of the codeword from a fixed codebook is used to convey additional important information. The decoder determines the multiplexing rule before decoding the rest of the data. The second method is based on partitioning a signal constellation into disjoint subsets in which the most important data sequence is encoded, using most of the available redundancy, to specify a sequence of subsets. The partitioning and code construction is done to maximize the minimum Euclidean distance between two different valid subset sequences. This leads to ways of partitioning the signal constellations into subsets. The less important data selects a sequence of signal points to be transmitted from the subsets. A side benefit of the proposed set partitioning procedure is a reduction in the number of nearest neighbors, sometimes even over the uncoded signal constellation.>

A least-squares model-based approach to digital halftoning is proposed. It exploits both a printer model and a model for visual perception. It obtains an optimal halftoned reproduction by minimizing the squared error between the response of the cascade of the printer and visual models to the binary image and the response of the visual model to the original gray-scale image. Least-squares model-based halftoning uses explicit eye models and relies on printer models that predict distortions and exploit them to increase, rather than decrease, both spatial and gray-scale resolution. The authors examine the one-dimensional case, in which each row or column of the image is halftoned independently. One-dimensional least-squares halftoning is implemented, in closed form, with the Viterbi algorithm. Experiments show that it produces better spatial and gray-scale resolution than conventional one-dimensional techniques and eliminates the problems associated with the modified (to account for printer distortions) error diffusion algorithm.>

The unequal error protection capabilities of convolutional codes belonging to the family of rate-compatible punctured convolutional codes (RCPC codes) are studied. The performance of these codes is analyzed and simulated for the first fading Rice and Rayleigh channels with differentially coherent four-phase modulation (4-DPSK). To mitigate the effect of fading, interleavers are designed for these unequal error protection codes, with the interleaving performed over one or two blocks of 256 channel bits. These codes are decoded by means of the Viterbi algorithm using both soft symbol decisions and channel state information. For reference, the performance of these codes on a Gaussian channel with coherent binary phase-shift keying (2-CPSK) is presented. A number of examples are provided to show that it is possible to accommodate widely different error protection levels within short information blocks. Unequal error protection codes for a subband speech coder are studied in detail. A detailed study of the effect of the code and channel parameters such as the encoder memory, the code rate, interleaver depth, fading bandwidth, and the contrasting performance of hard and soft decisions on the received symbols is provided.>

The problem of decoding data in the presence of infinite-duration intersymbol interference that is caused by severe channel filtering is considered. Filtered continuous phase modulations (CPM) are the particular object of study. A state-variable approach is used for defining the decoder tree. Maximum likelihood sequence estimation requires exhaustive tree searching, which is restricted by using the (M, L) algorithm since this approach does not require that the channel intersymbol interference be finite. After briefly describing the (M, L) algorithm, the authors motivate the problem of equalization of infinite-impulse-response channels by considering the performance of a discrete-time single-pole channel filtering a binary input sequence. The state variable description of a linear system is used to analyze the filtered modulation. The state of the filtered modulation for a given input modulation is used to define the tree structure of the filtered signal upon which the (M, L) algorithm operates. The minimum signal space distance results for several filtered CPM schemes are then summarized. Extensive simulation results are presented, and comparisons to the optimal performance are made.>

The coder is a dynamic bit allocation sub-band coder of the type first proposed by Ramstad (1982). It is structured in such a way that the relative importance of all bits is established as a byproduct of the dynamic bit allocation. It is shown that there is a difference in error sensitivity of four orders of magnitude between the most and the least important bits of the bit stream on average. A very flexible unequal error protection scheme is used to match the error sensitivity of the sub-band coder bit stream. This is accomplished using the concept of rate compatible punctured convolutional coding. The resulting coder gives robust performance over a simulated noisy channel. Its nominal bit rate is 16 kb/s, with 12 kb/s assigned for speech coding and 4 kb/s assigned for error correction. The entire encoder and decoder have been implemented on a single AT&T DSP-32 digital signal processor.>

Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processor.>

The minimum distance and hence the asymptotic error probability of trellis modulation codes that have been subjected to severe channel filtering are evaluated. A general algorithm is proposed for this purpose and applied to one class of coded modulations. Using a frequency-domain interpretation of the Euclidean distance, it is shown that under extreme channel band limitations, the error event that causes the minimum distance is characterized by a spectral null at DC. Numerical results show that for filtered continuous-phase-modulation codes, decreasing the modulation index or smoothing the frequency pulse does not improve the error performance. For some of these modulations, channel filtering can actually improve the minimum distance. Bandpass and low-pass filtering are contrasted. The numerical results are explained by power spectrum considerations.>

New group testing protocols are studied for a random multiple access, collision-type, slotted channel with binary feedback of either the success/failure or something/nothing variety. The protocols introduced here exhibit better performance than the ones suggested earlier. Implementation issues are considered.