Sidelink is a 3GPP-standardized technology that enables direct UE-to-UE communication. Initially designed for licensed 4G spectrum (Release 15), it expanded to 5G NR (Release 16, 17). However, due to the recent availability of large swaths of unlicensed spectrum (notably 6 GHz in U.S.), 3GPP has initiated studies to support sidelink operation in such unlicensed bands (denoted as SL-U). SL-U specifications ride on adaptation of 5G NR operation from licensed to unlicensed bands (NR-U). This paper provides a first analytical framework to quantify SL-U network performance in saturation conditions, for comparative evaluation to NR-U. A Markov chain model for NR-U is first presented, as a basis for necessary (significant) modifications needed to capture SL-U-specific features - the representation of gap states in the SL-U model to account for the restriction that SL-U transmissions only occur at SL slot boundaries. The resulting analysis highlights the differences and consequent (negative) impact of current 3GPP-defined SL-U channel access rulemaking; the analytical estimates are validated by results from a custom simulation of channel access. The primary takeaway is that the SL-U throughput under current 3GPP access rules suffers significantly relative to NR-U, and thus needs amelioration for successful marketplace adoption in the future.

This paper examines a direct-sequence code division multiple access (DS-CDMA) system for the reverse link in satellite communications on the move applications. In order to limit interference to adjacent satellites, strict limits on the effective isotropic radiated power (EIRP) spectral density are established by the Federal Communications Commission (FCC) and the International Telecommunications Union (ITU). Performance analysis of the DS-CDMA is carried out subject to this transmit power limitations and in the presence of channel and system impairments that are typically encountered in Ku-band (14-GHz band) satellite systems. Specific impairments considered are: small signal variations due to environmental conditions which are typically modeled by lognormal shadowing, atmospheric scintillations, and motion induced antenna pointing errors. Also, a novel methodology is developed to characterize motion induced antenna pointing errors and their impact on the system performance.

Future wireless multimedia terminals will have a variety of applications that require speech recognition capabilities. We consider a robust distributed speech recognition system where representative parameters of the speech signal are extracted at the wireless terminal and transmitted to a centralized automatic speech recognition (ASR) server. We propose two unequal error protection schemes for the ASR bit stream and demonstrate the satisfactory performance of these schemes for typical wireless cellular channels. In addition, a "soft-feature" error concealment strategy is introduced at the ASR server that uses "soft-outputs" from the channel decoder to compute the marginal distribution of only the reliable features during likelihood computation at the speech recognizer. This soft-feature error concealment technique reduces the ASR error rate by more than a factor of 2.5 for certain channels. Also considered is a channel decoding technique with source information that improves ASR performance.

A distributed automatic speech recognition (ASR) system is considered where features of the speech signal are extracted at the wireless terminal and transmitted to a centralized ASR server. An unequal error protection scheme is used for the quantized ASR feature stream. At the receiver, coherent demodulation is performed and the probability of error for each bit is computed using the max-log MAP algorithm. A 'soft-feature' decoding strategy is introduced at the ASR server that uses the marginal distribution of only the reliable features during likelihood computation. Alternatively, the confidence of each feature is computed from the bit error probabilities and each feature in the probability computation is weighted as a function of the feature confidence. The performance of the proposed soft-feature algorithms is evaluated over typical cellular wireless channels and it is shown to reduce ASR error rate by over 50% for certain channels at a small additional computational cost.

Future wireless multimedia terminals will have a variety of applications that require speech recognition capabilities. We consider a robust distributed speech recognition system where representative parameters of the speech signal are extracted at the wireless terminal and transmitted to a centralized automatic speech recognition (ASR) server. We propose several unequal error protection schemes for the ASR bit stream and demonstrate the satisfactory performance of these schemes for typical wireless cellular channels. In addition, a "soft-feature" error concealment strategy is introduced at the ASR server that uses "soft-outputs" from the channel decoder. This soft-feature error concealment techniques reduces the ASR error rate by up to four times for certain channels. Also considered is a channel decoding technique with source information that improves ASR performance.

High quality multimedia streaming applications over the Internet require very low packet loss rates. The Internet is characterized by long bursts of packet losses and delays. A large receive buffer can be used to mitigate the effects of packet losses and delays. However, a large receive buffer introduces a large delay in the playback of a packet. This large delay could be annoying at the start of a program or during switch over to another channel in a multi-channel broadcast. We introduce a separate low-delay tuning stream to address this start-up problem. In the steady state, this tuning stream is synchronized appropriately with the high-delay steady state stream to give an enhanced composite signal.

Several systems are being considered for digital audio broadcasting (DAB) and some of these systems will be deployed for commercial use. In this paper, we consider a code division multiplex (CDM) scheme for satellite DAB. For diversity purposes, this system makes use of at least two satellites, and in urban areas, where the satellite signals are shadowed by large structures, a terrestrial network is employed. We present a forward error protection (FEC) scheme that is robust under a variety of channel conditions, especially in cases where the received signal is severely attenuated because of highway underpasses and tunnels. This FEC scheme makes use of time diversity, and this introduces a large delay. Such delays will not affect the performance of broadcast systems; however, the delay will be an issue during tuning when a switch to another channel occurs. We introduce a separate low-delay low-rate tuning channel to facilitate tuning. Extensive simulation results are given to examine the performance of the system, and it is shown that about 70 audio channels, each with a rate of 96 kbit/s, can be satisfactorily supported in a bandwidth of approximately 12.5 MHz in most of the channel conditions considered.

Adaptive channel equalization accomplished without resorting to a training sequence is known as blind equalization. The Godard algorithm and the generalized Sato algorithm are two widely referenced algorithms for blind equalization of a QAM system. These algorithms exhibit very slow convergence rates when compared to algorithms employed in conventional data-aided equalization schemes. In order to speed up the convergence process, these algorithms may be switched over to a decision-directed equalization scheme once the error level is reasonably low. The authors present a scheme which is capable of operating in two modes: blind equalization mode and a mode similar to the decision-directed equalization mode. In this proposed scheme, the dominant mode of operation changes from the blind equalization mode at higher error levels to the mode similar to the decision-directed equalization mode at lower error levels. Manual switch-over to the decision-directed mode from the blind equalization mode, or vice-versa, is not necessary since transitions between the two modes take place smoothly and automatically.>

The authors consider a key algorithm for blind equalization and derive expressions for the evolution of the equalizer output error trajectory. The authors develop a model to examine the convergence behavior of this algorithm. Suitable approximations are incorporated into the model to facilitate analysis. The validity of these approximations is demonstrated for a typical communication channel. It is shown that even for a channel for which the assumption of Gaussianity of the equalizer input data may not be very good, the analysis presented predicts the convergence behavior reasonably well.>