Differential and coherent decorrelating multiuser receivers for space-time-coded CDMA systems

Previously, we proposed a differential space-code modulation (DSCM) scheme that integrates the strength of differential space-time coding and spreading to achieve interference suppression and resistance to time-varying channel fading in single-user environments. In this paper, we consider the problem of multiuser receiver design for code-division multiple-access (CDMA) systems that utilize DSCM for transmission. In particular, we propose two differential receivers for such systems. These differential receivers do not require the channel state information (CSI) for detection and, still, are resistant to multiuser interference (MUI) and time-varying channel fading. We also propose a coherent receiver that requires only the CSI of the desired user for detection. The coherent receiver yields improved performance over the differential receivers when reliable channel estimates are available (e.g., in slowly fading channels). The proposed differential/coherent receivers are decorrelative schemes that decouple the detection of different users. Both long and short spreading codes can be employed in these schemes. Numerical examples are presented to demonstrate the effectiveness of the proposed receivers.     

Adaptive source rate control for real‐time wireless video transmission

Hybrid ARQ schemes can yield much better throughput and reliability than static FEC schemes for the transmission of data over time-varying wireless channels. However these schemes result in extra delay. They adapt to the varying channel conditions by retransmitting erroneous packets, this causes variable effective data rates for current PCS networks because the channel bandwidth is constant. Hybrid ARQ schemes are currently being proposed as the error control schemes for real-time video transmission. An important issue is how to ensure low delay while taking advantage of the high throughput and reliability that these schemes provide for. In this paper we propose an adaptive source rate control (ASRC) scheme which can work together with the hybrid ARQ error control schemes to achieve efficient transmission of real-time video with low delay and high reliability. The ASRC scheme adjusts the source rate based on the channel conditions, the transport buffer occupancy and the delay constraints. It achieves good video quality by dynamically changing both the number of the forced update (intracoded) macroblocks and the quantization scale used in a frame. The number of the forced update macroblocks used in a frame is first adjusted according to the allocated source rate. This reduces the fluctuation of the quantization scale with the change in the channel conditions during encoding so that the uniformity of the video quality is improved. The simulation results show that the proposed ASRC scheme performs very well for both slow fading and fast fading channels. This revised version was published online in July 2006 with corrections to the Cover Date.     

An Enhanced A-MSDU Frame Aggregation Scheme for 802.11n Wireless Networks

The main goal of the IEEE 802.11n standard is to achieve a minimum throughput of 100 Mbps at the MAC service access point. This high throughput has been achieved via many enhancements in both the physical and MAC layers. A key enhancement at the MAC layer is frame aggregation in which the timing and headers overheads of the legacy MAC are reduced by aggregating multiple frames into a single large frame before being transmitted. Two aggregation schemes have been defined by the 802.11n standard, aggregate MAC service data unit (A-MSDU) and aggregate MAC protocol data unit (A-MPDU). As a consequence of the aggregation, new aggregation headers are introduced and become parts of the transmitted frame. Even though these headers are small compared to the legacy headers they still have a negative impact on the network performance, especially when aggregating frames of small payload. Moreover, the A-MSDU is highly influenced by the channel condition due mainly to lack of subframes sequence control and retransmission. In this paper, we have proposed an aggregation scheme (mA-MSDU) that reduces the aggregation headers and implements a retransmission control over the individual subframes at the MSDU level. The analysis and simulations results show the significance of the proposed scheme, specifically for applications that have a small frame size such as VoIP.     

A multi-CRC selective HARQ scheme for MIMO systems

A multi-Cyclic Redundancy Check （CRC） selective Hybrid Automatic-Repeat-reQuest （HARQ） scheme for improving the throughput efficiency of Multiple Input Multiple Output （MIMO） systems is proposed in this paper. According to different feedback information from the receiver, the proposed HARQ scheme employs two strategies, referred to as retransmission frame selection and space diversity. These two strategies decrease the successive frame errors upon retransmission. Theoretic analysis and computer simulation results show that this HARQ scheme achieves higher throughput than the existing HARQ schemes even in poor conditions of low Signal-to-Noise Ratio （SNR）.     

Two-step multipolling MAC protocol for wireless LANs

The IEEE 802.11 standard defines two coordination functions: distributed coordination function (DCF) and point coordination function (PCF). These coordination functions coordinate the shared wireless medium. The PCF uses a centralized polling-based channel access method to support time-bounded services. To design an efficient polling scheme, the point coordinator (PC) needs to obtain information about the current transmission status and channel condition for each station. To reduce overhead caused by polling frames, it is better to poll all stations using one polling frame containing the transmission schedule. In this paper, we propose an efficient polling scheme, referred to as two-step multipolling (TS-MP), for the PCF in wireless local area networks (WLANs). In this new scheme, we propose to use two multipolling frames with different purposes. The first frame is broadcast to collect information such as the numbers of pending frames and the physical-layer transmission rates for the communication links among all stations. The second frame contains a polling sequence for data transmissions designed based on the collected information. This frame is broadcast to all stations. Extensive simulation studies show that TS-MP not only overcomes the aforementioned deficiencies, but also help to implement rate adaptation over time-varying wireless channel.     

Erasure decoding in burst-error channels

Burst-error channels have been used to model a large class of modern communication media, and the problem of communicating reliably through such media has received much study [1]-[9]. Existing techniques include two-way communication schemes that involve error detection and retransmission, and schemes that utilize error correcting codes in code interleaving. The error-detection and retransmission scheme is simple, but its applicability has been restricted to limited environments. On the other hand, the concept of code interleaving has proved to be versatile and effective. Code interleaving distributes the error detection and correction burden among the component codes and thus lowers the overall redundancy requirement. However, the memory characteristics of the burst-error channel have not been used. This omission has prompted the investigation presented in this paper to utilize the inherent information embedded in the code interleaving scheme when used with burst-error channels. The concept of erasure decoding is introduced, leading to some useful coding and decoding strategies. Theoretical formulations are devised to predict code performance, and their validity is verified with computer simulations.     

Optimal Design of Mixed-Media Packet-Switching Networks: Routing and Capacity Assignment

This paper considers a mixed-media packet-switched computer communication network which consists of a low-delay terrestrial store-and-forward subnet combined with a low-cost high-bandwidth satellite subnet. We show how to route traffic via ground and/or satellite links by means of static, deterministic procedures and assign capacities to channels subject to a given linear cost such that the network average delay is minimized. Two operational schemes for this network model are investigated: one is a scheme in which the satellite channel is used as a slotted ALOHA channel; the other is a new multiaccess scheme we propose in which whenever a channel collision occurs, retransmission of the involved packets will route through ground links to their destinations. The performance of both schemes is evaluated and compared in terms of cost and average packet delay tradeoffs for some examples. The results offer guidelines for the design and optimal utilization of mixed-media networks.     

Performance study of link layer and MAC layer protocols to support TCP in 3G CDMA systems

Providing support for TCP with good quality link connection is a key issue for future wireless networks in which Internet access is going to be one of the most important data services. A number of schemes have been proposed in literature to improve the TCP performance over wireless links. In this paper, we study the performance of a particular combination of link layer protocol (e.g., radio link protocol or RLP) and MAC retransmissions to support the TCP connections over third generation (3G) wireless CDMA networks. We specifically investigate two metrics - the packet error rate and the delay provided by RLP and MAC retransmissions - both of which are important for TCP performance. For independent and identically distributed (i.i.d) error channels, we propose an analytical model for RLP performance with MAC retransmission. The segmentation of TCP/IP packets into smaller RLP frames, as well as the RLP buffering process, is modeled using a Markov chain. For correlated fading channels, we introduce an analytical metric called RLP retransmission efficiency. We show that: 1) the RLP frame size has significant impact on the overall 3G system performance, 2) MAC layer retransmissions significantly improve the TCP performance, and 3) the RLP retransmission scheme performs better in highly correlated channels, while other scheme performs better in low correlated channels. Simulation results also confirm these conclusions.     

Intercarrier interference in MIMO OFDM

In this paper, we examine multicarrier transmission over time-varying channels. We first develop a model for such a transmission scheme and focus particularly on multiple-input multiple output (MIMO) orthogonal frequency division multiplexing (OFDM). Using this method, we analyze the impact of time variation within a transmission block (time variation could arise both from Doppler spread of the channel and from synchronization errors). To mitigate the effects of such time variations, we propose a time-domain approach. We design ICI-mitigating block linear filters, and we examine how they are modified in the context of space-time block-coded transmissions. Our approach reduces to the familiar single-tap frequency-domain equalizer when the channel is block time invariant. Channel estimation in rapidly time-varying scenarios becomes critical, and we propose a scheme for estimating channel parameters varying within a transmission block. Along with the channel estimation scheme, we also examine the issue of pilot tone placement and show that in time-varying channels, it may be better to group pilot tones together into clumps that are equispaced onto the FFT grid; this placement technique is in contrast to the common wisdom for time-invariant channels. Finally, we provide numerical results illustrating the performance of these schemes, both for uncoded and space-time block-coded systems.     

Soft Information Combining for Turbo-MIMO Packet Retransmission

We studied three types of retransmission scheme for turbo-MIMO packet: Chase combining, incremental redundancy, and soft information combining, these three schemes are suitable for different situations. The MIMO channel in each retransmission is correlated in temporal dimension, and a standard method is utilized to simulate the retransmission channel model. Interleaving can shuffle the MIMO channel artificially, so the outage capacity of channel with interleaving is much better than the capacity without interleaving. If using different interleaver in retransmission, the receiver can only combine the retransmitted data after MIMO symbol demapping, we call it “soft information combing”. We find soft information combing is much useful in the true environment, we also find coding gain of incremental redundancy over Chase combining in most cases.

Selective Error Detection for Error-Resilient Wavelet-Based Image Coding

This paper introduces the concept of a similarity check function for error-resilient multimedia data transmission. The proposed similarity check function provides information about the effects of corrupted data on the quality of the reconstructed image. The degree of data corruption is measured by the similarity check function at the receiver, without explicit knowledge of the original source data. The design of a perceptual similarity check function is presented for wavelet-based coders such as the JPEG2000 standard, and used with a proposed ldquoprogressive similarity-based ARQrdquo (ProS-ARQ) scheme to significantly decrease the retransmission rate of corrupted data while maintaining very good visual quality of images transmitted over noisy channels. Simulation results with JPEG2000-coded images transmitted over the binary symmetric channel, show that the proposed ProS-ARQ scheme significantly reduces the number of retransmissions as compared to conventional ARQ-based schemes. The presented results also show that, for the same number of retransmitted data packets, the proposed ProS-ARQ scheme can achieve significantly higher PSNR and better visual quality as compared to the selective-repeat ARQ scheme.     

Rate-adaptive transmission over correlated fading channels

In this paper, we investigate link adaptation and incremental redundancy (IR) retransmission schemes over correlated wireless channels. While computer simulations have been used to study the performance of these techniques, a numerically tractable analytical approach is more desirable to analyze generic protocols, and to reveal insights into the performance tradeoffs. An error-recursion approach is developed in this paper to mathematically analyze the throughput, delay, and energy efficiency of rate-adaptation techniques over fading channels with arbitrary correlations between retransmissions. Using Reed-Solomon codes as an example, we quantitatively predict the performance tradeoff of throughput and latency for IR schemes and the performance dependency on the channel correlation. Numerical results also show that reactive rate-adaptation schemes with IR retransmission outperform proactive rate-adaptive schemes, even with perfect channel side information, in terms of throughput and energy efficiency.     

Capacity of Time-Varying Channels With Causal Channel Side Information

We derive the capacity of time-varying channels with memory that have causal channel side information (CSI) at the sender and receiver. We obtain capacity of block-memoryless and asymptotically block-memoryless channels with block-memoryless or weakly decorrelating side information. Our coding theorems rely on causal generation of the codewords relative to the causal transmitter CSI. The CSI need not be perfect, and we consider the case where the transmitter and receiver have the same causal CSI as well as the case where the transmitter CSI is a deterministic function of the receiver CSI. For block-memoryless and asymptotically block-memoryless channels, our coding strategy averages mutual information density over multiple transmission blocks to achieve the maximum average mutual information. We apply the coding theorem associated with the block-memoryless channel to determine the capacity and optimal input distribution of intersymbol interference (ISI) time-varying channels with causal perfect CSI about the time-varying channel. The capacity of this channel cannot be found through traditional decomposition methods     

Bit Interleaved Time-Frequency Coded Modulation for OFDM Systems Over Time-Varying Channels

Bit Interleaved Time-Frequency Coded Modulation for OFDM Systems Over Time-Varying Channels Orthogonal frequency-division multiplexing (OFDM) is a promising technology in broadband wireless communications with its ability in transforming a frequency selective fading channel into multiple flat fading channels. However, the time-varying characteristics of wireless channels induce the loss of orthogonality among OFDM sub-carriers, which was generally considered harmful to system performance. In this paper, we propose a bit interleaved time–frequency coded modulation (BITFCM) scheme for OFDM to achieve both time and frequency diversity inherent in broadband time-varying channels. We will show that the time-varying characteristics of the channel are beneficial to system performance. Using the BITFCM scheme and for relatively low maximum normalized Doppler frequency, a reduced complexity Maximum Likelihood (ML) decoding approach is proposed to achieve good performance with low complexity as well. For high maximum normalized Doppler frequency, the inter-carrier interference (ICI) can be large and an error floor will be induced. To solve this problem, we propose two ICI mitigation schemes by taking advantage of the second order channel statistics and the complete channel information, respectively. It will be shown that both schemes can reduce the ICI significantly.     

Channel capacity of fading channels for differentially encodedtransmission

The capacity of flat fading channels when applying differential encoding with noncoherent reception and no channel state information available at the receiver is considered. Numerical results indicate the gains achievable by multiple symbol detection in the case of slowly time-varying channels and provide a comparison between schemes with different potential bandwidth efficiencies     

Robust image transmission over energy-constrained time-varyingchannels using multiresolution joint source-channel coding

We explore joint source-channel coding (JSCC) for time-varying channels using a multiresolution framework for both source coding and transmission via novel multiresolution modulation constellations. We consider the problem of still image transmission over time-varying channels with the channel state information (CSI) available at (1) receiver only and (2) both transmitter and receiver being informed about the state of the channel, and we quantify the effect of CSI availability on the performance. Our source model is based on the wavelet image decomposition, which generates a collection of subbands modeled by the family of generalized Gaussian distributions. We describe an algorithm that jointly optimizes the design of the multiresolution source codebook, the multiresolution constellation, and the decoding strategy of optimally matching the source resolution and signal constellation resolution “trees” in accordance with the time-varying channel and show how this leads to improved performance over existing methods. The real-time operation needs only table lookups. Our results based on a wavelet image representation show that our multiresolution-based optimized system attains gains on the order of 2 dB in the reconstructed image quality over single-resolution systems using channel optimized source coding     

Turbo Equalization with Prediction and Iterative Estimation of Time-Varying Frequency-Selective Channels

Turbo equalization that cooperates with channel prediction and iterative channel estimation is investigated for mobile wireless communications. Frames of information bits are encoded, interleaved, and mapped to symbols for transmission over time-varying frequency-selective fading channels. At the receiver, the Turbo equalizer consists of a maximum a posteriori probability equalizer/demapper and a soft-input soft-output maximum a posteriori probability decoder. With initial channel estimates and sparse pilot insertion across a number of frames, the receiver predicts the channel of the current frame. The effect of error propagation of channel prediction is mitigated by the de-interleaver that is embedded in the Turbo equalizer. The predicted and interpolated channel is refined through a channel estimator that uses the soft estimates of data symbols at each Turbo iteration. Due to the bandlimiting feature of channel variation, the channel estimation error can be smoothed by low-pass filters that follow the channel estimator. Simulation results show that incorporating Turbo equalization with channel prediction and iterative channel estimation can combat time- and frequency-selective fading and improve reception performance.     

Aggregation With Fragment Retransmission for Very High-Speed WLANs

In upcoming very high-speed wireless LANs (WLANs), the physical (PHY) layer rate may reach 600 Mbps. To achieve high efficiency at the medium access control (MAC) layer, we identify fundamental properties that must be satisfied by any CSMA-/CA-based MAC layers and develop a novel scheme called aggregation with fragment retransmission (AFR) that exhibits these properties. In the AFR scheme, multiple packets are aggregated into and transmitted in a single large frame. If errors happen during the transmission, only the corrupted fragments of the large frame are retransmitted. An analytic model is developed to evaluate the throughput and delay performance of AFR over noisy channels and to compare AFR with similar schemes in the literature. Optimal frame and fragment sizes are calculated using this model. Transmission delays are minimized by using a zero-waiting mechanism where frames are transmitted immediately once the MAC wins a transmission opportunity. We prove that zero-waiting can achieve maximum throughput. As a complement to the theoretical analysis, we investigate the impact of AFR on the performance of realistic application traffic with diverse requirements by simulations. We have implemented the AFR scheme in the NS-2 simulator and present detailed results for TCP, VoIP, and HDTV traffic. The AFR scheme described was developed as part of the IEEE 802.11n working group work. The analysis presented here is general enough to be extended to proposed schemes in the upcoming 802.11n standard. Trends indicated in this paper should extend to any well-designed aggregation schemes.