On Channel Modelling for 30 (WCDMA) Systems Using Turbo-Coding

Channel coding is a fundamental procedure in wireless telecommunication systems and has a strong impact on the data transmission quality. This effect becomes more important when the transmission must be characterised by low latency and low bit error rate, as in the case of vehicle-to-everything (V2X) services. Thus, V2X services must use powerful and efficient coding schemes. In this paper, we thoroughly examine the performance of the most important channel coding schemes in V2X services. More specifically, the impact of use of 4th-Generation Long-Term Evolution (4G-LTE) turbo codes, 5th-Generation New Radio (5G-NR) polar codes and low-density parity-check codes (LDPC) in V2X communication systems is researched. For this purpose, we employ stochastic propagation models that simulate the cases of line of sight (LOS), non-line of sight (NLOS) and line of sight with vehicle blockage (NLOSv) communication. Different communication scenarios are investigated in urban and highway environments using the 3rd-Generation Partnership Project (3GPP) parameters for the stochastic models. Based on these propagation models, we investigate the performance of the communication channels in terms of bit error rate (BER) and frame error rate (FER) performance for different levels of signal to noise ratio (SNR) for all the aforementioned coding schemes and three small V2X-compatible data frames. Our analysis shows that turbo-based coding schemes have superior BER and FER performance than 5G coding schemes for the vast majority of the considered simulation scenarios. This fact, combined with the low-complexity requirements of turbo schemes for small data frames, makes them more suitable for small-frame 5G V2X services.

By dividing the available spectrum into multiple bands, multi-band ultra-wideband (UWB) systems possess many advantages, such as the flexibility to eliminate possible interference and coexistence problems with available wireless standards. UWB channels have large delay spread, causing inter- symbol interference (ISI) in high rate systems. The RAKE receiver is often used to capture multipath energy, but it is incapable of ISI compensation, causing severe bit error rate (BER) degradation in practical UWB channels. This paper proposes an over-sampling multi-channel equalizer per sub-band for multi-band UWB. The performance of the proposed receiver is compared with the RAKE receiver through analytical and simulation studies. The proposed equalizer has been shown to be able to handle ISI under any channel conditions or data rate presented, with an acceptable BER. In addition, the rich multipath diversity of the UWB channels is harnessed by the over-sampling scheme, for output SNR improvement. one. Therefore, equalization receiver based on adaptive algorithms seems to be an effective way to recover the transmitted data. With this consideration, we propose an over-sampling multi-channel equalizer per sub-band for a multi-band UWB system, which provides more effective inter-symbol interference (ISI) suppression compared with the conventional RAKE receiver when operating in channels with large delay spread. Through detailed theoretical analysis and extensive simulations, we are able to show that the proposed 4-arm equalizer scheme is superior to a 4-arm conventional RAKE receiver in the presence of ISI at all signal-to-noise ratios (SNRs). Based on the method presented in (4), the simulated performance is shown to conform to analytical predictions. The RAKE receiver is incapable of ISI compensation, and a moderate level of ISI is shown to cause severe bit error rate (BER) degradation. The proposed over-sampling equalizer, on the other hand, is able to handle ISI under any channel conditions or data rate, with an acceptable BER. Over- sampling is shown to be able to achieve diversity gain, and improves the output SNR.

The use of open-loop coding can be easily extended to a closed-loop concatenated code if the transmitter has access to feedback. This can be done by introducing a feedback transmission scheme as an inner code. In this paper, this process is investigated for the case when a linear feedback scheme is implemented as an inner code and, in particular, over an additive white Gaussian noise (AWGN) channel with noisy feedback. To begin, we look to derive an optimal linear feedback scheme by optimizing over the received signal-to-noise ratio (SNR). From this optimization, a linear feedback scheme is produced that is asymptotically optimal in the sense of blocklength-normalized SNR; it is then compared to other well-known schemes. Then, the linear feedback scheme is implemented as an inner code to a concatenated code over the AWGN channel with noisy feedback. This code shows improvements not only in error exponent bounds, but also in bit error rate (BER) and frame error rate (FER). It is also shown that if the concatenated code has total blocklength L and the inner code has blocklength, N, the inner code blocklength should scale as N = O(C/R), where C is the capacity of the channel and R is the rate of the concatenated code. Simulations with low-density parity-check (LDPC) and turbo codes are provided to display practical applications and their error rate benefits.

Recently, low-density parity-check (LDPC) codes have attracted much attention especially in mobile communication system as a good simple error correcting codes achieving the data throughput near Shannon limit with lower complexity. In this paper, we design some regular LDPC codes with very low bit error rate (BER) and frame error rate (FER) according to the downlink parameters in TD-SCDMA system. We construct the downlink physical layer emulation model with matlab according to TD-SCDMA system's baseband signal disposing flow chart, and execute the link simulation with the performance analysis for some available channel codes and the designed regular LDPC codes based on the proposed emulation model. Passing the simulation, we obtain some conclusions that the regular LDPC codes are better than the convolutional codes or turbo codes in error rate performance.

MIMO systems (with multiple transmit and receive antennas) are becoming increasingly popular, and many next-generation systems such as WiMAX, 3-GPP LTE and IEEE802.11 n wireless LANs rely on the increased throughput of MIMO systems with up to four antennas at receiver and transmitter. High throughput implementation of the detection unit for MIMO systems is a significant challenge. This challenge becomes still harder, because the above mentioned standards demand support for multiple modulation and coding schemes. This implies that the MIMO detector must be dynamically reconfigurable. Also, to achieve required bit error rate (BER) or frame error rate (FER) performance, the detector has to provide soft values to advanced forward error correction (FEC) schemes like turbo Codes. This paper presents an ASIC implementation of a novel MIMO detector architecture that is able to reconfigure on the fly and provides soft values as output. The design is implemented in 45 nm predictive technology library, and has a parallelism factor of four. The detector has many qualities of a systolic architecture and achieves a continuous throughput of 1 Gbps for QPSK, 500 Mbps for 16-QAM, and 187.5 Mbps for 64-QAM. The total area is estimated to be approximately 70 KGates equivalent, and power consumption is estimated to be 114 mW.

MIMO systems (with multiple transmit and receive antennas) are becoming increasingly popular, and many next-generation systems such as WiMAX, 3-GPP LTE and IEEE802.11n wireless LANs relay on the increased throughput of MIMO systems with up to four antennas at receiver and transmitter. High throughput implementation of the detection unit for MIMO systems is a significant challenge especially for higher order modulation schemes. To achieve superior Bit Error Rate (BER) or Frame Error Rate (FER) performance, the detector has to provide soft values to advanced Forward Error Correction (FEC) schemes like Turbo Codes. This paper presents a systolic soft detector architecture for high dimensional (eg. 4times4, 64-QAM) MIMO systems. A Single detector core achieves, throughput of 215 Mbps and power consumption of 23.6 mW, whiles using only 33.1 K gate equivalent (for l <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> norm). Impressive SNR gains of almost 2 dB are observed with respect to the hard detection counterpart over a block fading channel (at an FER of 1%). Additionally, the architecture can be stacked to give linear increase in throughput with linear increase in hardware resources.

In this paper we analyze the performance in terms of Bit Error Rate (BER) and Frame Error Rate (FER) of a duobinary turbo coded system over Rayleigh flat fading channels. The turbo codes (TCs) proposed here are constructed from parallel concatenation of two identical rate 2/3 recursive systematic convolutional RSC codes. It results rate 1/2 DuoBinary Turbo Codes (DBTCs). Data block length has k=m×N=2×752 bits. We consider the family of interleaver proposed in 2005 by C. Douillard and C. Berrou which is simpler to implement in hardware. The results from this paper show that the two DBTC found by us, outperforms the DBTC proposed in literature with more than 0.2 dB for a BER<;10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-6</sup> and a FER of 10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-4</sup> , also when a Rayleigh channel is considered.

A new type of interleaver, block-random interleaver, is proposed and compared with the principal types of interleavers on the basis of the BER (bit error rate) performances of the turbo code. A recursive systematic convolutional code, 1/3 rate, unpunctured turbo code was used. The convolutional code has the constraint length, K=4 with generator matrix [1, 15/13], in octal representation. The MAP algorithm, 12 iterations were used. A log likelihood ratio (LLR) stop criterion was selected. The BER and FER (frame error rate) for different interleaver lengths were obtained.

This paper presents a novel and cost effective method to be used in the optimization of the Gaussian Frequency Shift Keying (GFSK) at the receiver of the Bluetooth communication system. The proposed method enhances the performance of the noncoherent demodulation schemes by improving the Bit Error Rate (BER) and Frame Error Rate (FER) outcomes. Linear, Extended, and Unscented Kalman Filters are utilized in this technique. A simulation model, using Simulink, has been created to simulate the Bluetooth voice transmission system with the integrated filters. Results have shown improvements in the BER and FER, and that the Unscented Kalman Filters (UKF) have shown superior performance in comparison to the linear Kalman Filter (KF) and the Extended Kalman Filter (EKF). To the best of our knowledge, this research is the first to propose the usage of the UKF in the optimization of the Bluetooth System receivers in the presence of additive white Gaussian noise (AWGN), as well as interferences.

Increasing the performances of turbo codes used in communication systems is an important issue at the moment. The method used for trellis termination influences the code performances. This paper describes and compares (in terms of bit error rate - BER and frame error rate - FER) four types of these methods: without trellis termination, termination only of the first trellis, post-interleaver flushing and dual termination. Turbo code performances are evaluated for two lengths of S-random interleaver, using binary phase-shift keying (BPSK) modulation, over Additive White Gaussian Noise (AWGN) channel. The simulations were performed for two generating matrices of the component encoders and the decoding algorithm we have used is a version of Max-Log-MAP.

More advanced receiver structures than the conventional single antenna Rake can be used to improve the signal-to-noise (SNR) ratios, which is especially beneficial in order to utilize the high bit rates provided by the HSDPA concept in Wideband Code Division Multiple Access (WCDMA) network. In WCDMA system, orthogonal Walsh---Hadamard sequences are used as channelization codes. In frequency-selective fading channels the orthogonality of channelization codes disappears and intra-cell multiple access interference (MAI) arises. In order to mitigate the effect of MAI, chip-level equalization has shown to be a simple and effective solution. The effectiveness of chip equalization, however, degrades at the cell borders where the inter-cell interference dominates rather than MAI. Dual antenna reception is a straight-forward solution to mitigate that performance drop. In this paper, we present an analysis of the expected gains of advanced receivers over conventional single antenna Rake receiver in realistic situations by using a dynamic WCDMA system-level tool. Considered advanced receivers include single and dual antenna Linear Minimum Mean Squared Error (LMMSE) chip-level equalizers and dual antenna Rake receiver. The network performance with advanced receivers is studied also from a more practical point of view by assuming that the penetration of advanced HSDPA terminal receivers is gradually increased in the network.

In order to solve the high bit error rate (BER) caused by rate oscillation in traditional Data Rate Change (DRC) algorithm, an improved DRC algorithm based on Adaptive Frame Length (AFL) was proposed. Firstly, the frame length and transmission rate of the initial transmission were determined by the parameters of the current channel and the information of previous empirical values. Then, if two frames with the same length were successively sent in the transmission process, the frame length would be accordingly increased. If the retransmission failed twice in a row, the frame length would be halved in the next transmission. Finally, the frame error rate was calculated based on the current frame length. The data rate would be updated If the value was less than the present threshold. Compared with RapidM DRC, the average bit error rate of the link decreased by 4.8 percentage points, and the link availability increased by 10 percentage points. The experimental results show that the proposed algorithm can eliminate the rate oscillation and improve the communication capability of the HF communication system.

This paper presents the performance of a hard decision belief propagation (HDBP) decoder used for Luby transform (LT) codes over additive white Gaussian noise channels; subsequently, three improved HDBP decoders are proposed. We first analyze the performance improvement of the sorted ripple and delayed decoding process in a HDBP decoder; subsequently, we propose ripple-sorted belief propagation (RSBP) as well as ripple-sorted and delayed belief propagation (RSDBP) decoders to improve the bit error rate (BER). Based on the analysis of the distribution of error encoded symbols, we propose a ripple-sorted and threshold-based belief propagation (RSTBP) decoder, which deletes low-reliability encoded symbols, to further improve the BER. Degree distribution significantly affects the performance of LT codes. Therefore, we propose a method for designing optimal degree distributions for the proposed decoders. Through simulation results, we demonstrate that the proposed RSBP and RSDBP decoders provide significantly better BER performances than the HDBP decoder. RSDBP and RSTDP combined with the proposed degree distributions outperformed state-of-the-art degree distributions in terms of the number of encoded symbols required to recover an input symbol correctly (NERRIC) and the frame error rate (FER). For a hybrid decoder formulated by combining RSDBP with a soft decision belief propagation decoder, the proposed degree distribution outperforms the other degree distributions in terms of decoding complexity.

This paper addresses the design of semi-random, prunable interleavers for parallel concatenated convolutional codes (PCCC). The proposed technique is iterative and is based on the growth of a smaller interleaver up to the desired length N. The optimization is achieved via a minimization using a cost-function strictly related to both the correlation properties of the extrinsic information and the concept of spread of an interleaver. Performance of the designed interleavers are given in terms of bit error rate (BER) and frame error rate (FER). Comparisons are given with respect to other prunable and ad-hoc interleaver design techniques already proposed in literature. The designed interleavers are prunable and have a behavior very similar to the interleavers designed with techniques which maximize the spread of the permutation

Turbo codes are proposed in most of the advanced digital communication standards, such as 3GPP-LTE. However, due to its computational complexity, the turbo decoder is one of the most power hungry blocks in digital baseband. To alleviate this issue, one way is to avoid surplus computing phases thanks to the early termination of the iterative decoding process. The use of stopping criteria is one of the most common algorithm level power reduction methods in literature. These methods always come with some hardware overhead. In this paper, a new trellis based stopping criterion is proposed. The novelty of this approach is the lower hardware overhead thanks to the use of trellis states as key parameter to stop the iterative process. Results are showing the importance of this added hardware in terms of method efficiency. Compared to state-of-the-art Log Likelihood Ratio (LLR) based techniques, proposed Low Complexity Trellis Based (LCTB) is demonstrating 23% less power consumption on average, for comparable performance level in terms of Bit Error Rate (BER) and Frame Error Rate (FER).