We propose a demodulation framework to extend the maximum distance of unrepeated transmission systems, where the simplest back propagation (BP), polarization and phase recovery, data arrangement for machine learning (ML), and symbol decision based on ML are rationally combined. The deterministic waveform distortion caused by fiber nonlinearity and chromatic dispersion is partially eliminated by BP whose calculation cost is minimized by adopting the single-step Fourier method in a pre-processing step. The non-deterministic waveform distortion, i.e., polarization and phase fluctuations, can be eliminated in a precise manner. Finally, the optimized ML model conducts the symbol decision under the influence of residual deterministic waveform distortion that cannot be cancelled by the simplest BP. Extensive numerical simulations confirm that a DP-16QAM signal can be transmitted over 240km of a standard single-mode fiber without optical repeaters. The maximum transmission distance is extended by 25km.

With the advent of the Internet of Things (IoT), short packet transmissions will dominate the future wireless communication. However, traditional coherent demodulation and channel estimation schemes require large pilot overhead, which may be highly inefficient for short packets in multipath fading scenarios. This paper proposes a novel pilot-free short packet structure based on the association of modulation on conjugate-reciprocal zeros (MOCZ) and tail-biting convolutional codes (TBCC), where a noncoherent demodulation and decoding scheme is designed without the channel state information (CSI) at the transceivers. We provide a construction method of constellation sets and demodulation rule for M-ary MOCZ. By deriving low complexity log-likelihood ratios (LLR) for M-ary MOCZ, we offer a reasonable balance between energy and bandwidth efficiency for joint coding and modulation scheme. Simulation results show that our proposed scheme can attain significant performance and throughput gains compared to the pilot-based coherent modulation scheme over multipath fading channels.

Laser Doppler Vibrometers (LDVs) enable the acquisition of remote speech signals by measuring small-scale vibrations around a target. They are now widely used in the fields of information acquisition and national security. However, in remote speech detection, the coherent measurement signal is subject to environmental noise, making detecting and reconstructing speech signals challenging. To improve the detection distance and speech quality, this paper proposes a highly accurate real-time speech measurement method that can reconstruct speech from noisy coherent signals. First, the I/Q demodulation and arctangent phase discrimination are used to extract the phase transformation caused by the acoustic vibration from coherent signals. Then, an innovative smoothness criterion and a novel phase difference-based dynamic bilateral compensation phase unwrapping algorithm are used to remove any ambiguity caused by the arctangent phase discrimination in the previous step. This important innovation results in the highly accurate detection of phase jumps. After this, a further innovation is used to enhance the reconstructed speech by applying an improved waveform-based linear prediction coding method, together with adaptive spectral subtraction. This removes any impulsive or background noise. The accuracy and performance of the proposed method were validated by conducting extensive simulations and comparisons with existing techniques. The results show that the proposed algorithm can significantly improve the measurement of speech and the quality of reconstructed speech signals. The viability of the method was further assessed by undertaking a physical experiment, where LDV equipment was used to measure speech at a distance of 310m in an outdoor environment. The intelligibility rate for the reconstructed speech exceeded 95%, confirming the effectiveness and superiority of the method for long-distance laser speech measurement.

This paper proposes individual computation processes of the partial demodulation reference signal (DM-RS) sequence in a synchronization signal (SS)/physical broadcast channel (PBCH) block to be used to detect the radio frame timing based on SS/PBCH block index detection for New Radio (NR) initial access. We present the radio frame timing detection probability using the proposed partial DM-RS sequence detection method that is applied subsequent to the physical-layer cell identity (PCID) detection in five tapped delay line (TDL) models in both non-line-of-sight (NLOS) and line-of-sight (LOS) environments. Computer simulation results show that by using the proposed method, the radio frame timing detection probabilities of almost 100% and higher than 90% are achieved for the LOS and NLOS channel models, respectively, at the average received signal-to-noise power ratio (SNR) of 0dB with the frequency stability of a local oscillator in a set of user equipment (UE) of 5ppm at the carrier frequency of 4GHz.

We propose a spectrum regeneration and demodulation method for multiple direct RF undersampled real signals by using a new algorithm. Many methods have been proposed to regenerate the RF spectrum by using undersampling because of its simple circuit architecture. However, it is difficult to regenerate the spectrum from a real signal that has a band wider than a half of the sampling frequency, because it is difficult to include complex conjugate relation of the folded spectrum into the linear algebraic equation in this case. We propose a new spectrum regeneration method from direct undersampled real signals that uses multiple clocks and an extended algorithm considering the complex conjugate relation. Simulations are used to verify the potential of this method. The validity of the proposed method is verified by using the simulation data and the measured data. We also apply this algorithm to the demodulation system.

The conventional Equal Gain Transmission and Maximum Ratio Combining (EGT/MRC) requires nonlinear optimization to find the optimal beamforming vector at the receiver. This study shows that the optimal beamforming vector can be easily formed by the geometrical concepts. Accordingly, a novel transmission/reception scheme, called the Scalar Equal Gain Transmission and Generalized Maximum Ratio Combining (SEGT/GMRC), is presented and examined. The Monte-Carlo simulations validate the theory and it is shown that the optimal beamforming vector formed by SEGT is the same as the one determined by the nonlinear optimizer. The closed-form analytical error performance of the SEGT/GMRC scheme is also derived for multiple input single output (MISO) communications. This study also introduces the new limited-feedback geometrical codebooks, called the Quantized Equal gain (QE) codebooks, which can be easily installed as symbol mappers. These codebooks are based on quantized SEGT/GMRC, which eliminates the need for any iterative searching scheme, such as exhaustive search at the receiver. The minimum amount of feedback bits depends on the modulation scheme, where a general M-PSK modulation requires at least log2M bits per quantized phase angle. It is also shown that BPSK modulation requires at least 2 bits per quantized phase angle for near-optimal performance.

On rocket-borne GPS receivers, the high dynamics make the use of PLL is impractical and a FLL is used. This in turn leads to a string of errors on the demodulated navigation message. The present paper proposes two decoding algorithms for GPS navigation messages suited for strings of errors. The first philosophy uses parity check syndromes to correct demodulation errors. The second philosophy uses Log Correlation Ratio (LCR) comparison along with parity check syndromes to correct for the most probable error pattern. GPS data availability can be significantly improved by using the latter.

In this paper, we propose a soft-decoding near-ML MIMO demodulation scheme that achieves near optimal performance with fixed and low complexity. Exploiting the regular structure of bit-to-symbol mapping, the proposed scheme performs hard demodulation to find the first candidate symbol for each stream followed by selection of nearby candidate points such that at least one candidate exists for the computation of likelihood information of bit 0 and 1 without intermediate calculation of the Euclidean distance. This demodulation scheme enables an improvement in performance by guaranteeing the existence of candidates and a significant reduction in the number of distance calculations which is a major complexity burden. The performance is evaluated by computer simulation, and computational complexity is also assessed in terms of the number of complex multiplication.

This letter presents the architecture of multi-gigabit parallel demodulator suitable for demodulating high order QAM modulated signal and easy to implement on FPGA platform. The parallel architecture is based on frequency domain implementation of matched filter and timing phase correction. Parallel FIFO based delete-keep algorithm is proposed for timing synchronization, while a kind of reduced constellation phase-frequency detector based parallel decision feedback PLL is designed for carrier synchronization. A fully pipelined parallel adaptive blind equalization algorithm is also proposed. Their parallel implementation structures suitable for FPGA platform are investigated. Besides, in the demonstration of 2 Gbps demodulator for 16QAM modulation, the architecture is implemented and validated on a Xilinx V6 FPGA platform with performance loss less than 2 dB.

We demonstrate phase demodulation of 10-Gbps DPSK signals using a silicon micro-ring resonator with a radius of 10 µm and with various coupling gaps for light of ∼1550 nm in wavelength. Influence of the Q factors and transmissions of the resonators on the response speed and power balance of the two output ports is discussed. Furthermore, temperature sensitivity on resonance peak was measured and we discuss its effect on practical demodulation application.

In this paper, a joint blind synchronization and demodulation scheme is developed for ultra-wideband (UWB) impulse radio systems. Based on the prior knowledge of the direct-sequence (DS) spread codes, the proposed approach can achieve frame-level synchronization with the help of frame-rate samples. Taking advantage of the periodicity of the DS spread codes, the frame-level synchronization can be carried out even in one symbol interval. On the other hand, after timing acquisition, these frame-rate samples can be re-utilized also for demodulation. Thus the acquisition time and the implementation complexity are reduced considerably. The performance improvement can be justified by both theoretical analysis and simulation results, in terms of acquisition probability and bit error rate (BER).

In OFDMA systems, various subcarrier allocation (SA) algorithms have been developed and adopted to realize the low-cost implementation or the optimized usage of resources, such as bandwidth and total transmit power, at the cost of increased complexity. Regardless of SA algorithms, however, it is computationally inefficient for a user who uses only a small number of subcarriers to use a full fast Fourier transform (FFT) for multicarrier demodulation at a conventional receiver in a downlink environment. In response, this letter proposes a novel low complexity FFT scheme that demodulates a set of desired multicarriers with smaller-size FFTs is proposed for computationally efficient and/or low hardware-cost receiver in OFDMA systems. Furthermore, a decision rule for the optimum size of FFT in the known transform decomposition methods is provided.

The system under study is a convolutionally coded and orthogonally modulated DS-CDMA system over time-varying frequency-selective Rayleigh fading channels in multiuser environments. Iterative soft demodulation and decoding using the Turbo principle can be applied to such a system to increase the system capacity and performance. To combat multiple access interference (MAI), we incorporate the interference cancellation (IC) and decision-directed channel estimation (CE) in the demodulator. However, both IC and CE are subject to performance degradation due to incorrect decisions. In order to prevent error propagation from the decision feedback, soft interference cancellation and channel estimation assisted demodulation is proposed in this paper. The performance of this strategy is evaluated numerically and proved to be superior to the hard decision-directed approach with a minor increase in complexity.

This letter describes an efficient architecture for a Software Defined Radio (SDR) Wideband Code Division Multiple Access (WCDMA) receiver using for high performance wireless communication systems. The architecture is composed of a Radio Frequency (RF) front-end, an Analog-to-Digital Converter (ADC), and a Quadrature Amplitude Modulation (QAM) demodulator. A coherent demodulator, with a complete digital synchronization scheme, achieves the bit-error rate (BER) of 10-6 with the implementation loss of 0.5 dB for a raw Quadrature Phase Shift King (QPSK) signal.

A directional audible sound can be generated by amplitude-modulated (AM) into ultrasound wave from a parametric array. To synthesize audio signals produced by the self-demodulation effect of the AM sound wave, a quasi-linear analytical solution, which describes the nonlinear wave propagation, is developed for fast numerical evaluation. The radiated sound field is expressed as the superposition of Gaussian Beams. Numerical results are presented for a rectangular parametric loudspeaker, which are in good agreement with the experimental data published previously.

The objective of this paper is to study the relationship between a visual stimulus and the amplitude and phase of the alpha wave as a first step to investigating a change in the background wave after a sensory stimulus and an evoked potential. We examined the effect of a single visual stimulus on the amplitude and phase of alpha waves using the complex demodulation method. The visual stimuli were generated by an LED mounted in goggles with the eyes-closed condition. The amplitude of the alpha wave decreased gradually after the stimulus, until it reached a minimum at around 300 ms after the stimulus. The alpha wave continued to increase, showing some rebound, and returning again to the pre-stimulus level. The phase variation after the stimulus tends to be considerably larger than that before the stimulus. Moreover, the average phase returned to the same slope as the pre-stimulus by 2550 ms after the stimulus. The visual stimulus has an effect on the alpha wave until about 2500 ms after the stimulus. The phase variation difference before and after stimulus is significant from 112 ms to 678 ms after the stimulus. This finding suggests there is a partially pararell time course between the change in VEPs plus ERP complex and the alpha wave.

A new analog correlator circuit is proposed for direct sequence code division multiple access (DS-CDMA) demodulator. The circuit consists of only 16 switches, 4 capacitors and 2 level shifters. Control sequence requires only three clock phases. Simulation with code length of 127 reveals that the proposed circuit has a good ability to cancel off the charge error and dissipates 3.4mW at 128MHz. The circuit had been designed using a 0.6µm CMOS process. The area of 256µm 245µm is estimated to be 9 times smaller compared to other reported equivalent analog correlators.

This paper proposes a new blind adaptive algorithm for computing the weight vector of an antenna array system. The new technique utilizes a Generalized On-Off algorithm to obtain the weight vector from the pilot channel of IS2000 1X system, of which the processing gain can be controlled arbitrarily. The proposed algorithm generates a suboptimal weight vector maximizing the SINR (signal to interference plus noise ratio) with a linear computational load. Based on the analysis obtained from various simulations, it is observed that the proposed algorithm is suitable for the practical IS2000 1X mobile communication environments.

Active integrated antennas are a maturing topic. Many novel configurations have been described and system designers are how investigating how the advantages of compactness and increased functionality can be exploited in applications. In this paper, the various types of integrated antennas are discussed together with possible ways of exploiting the technology. New configurations of direct conversion integrated antennas are then described in detail, which illustrate some of the possibilities inherent in the technology.

Upper bounds on the bit error rate (BER) for maximum likelihood (ML) decoding are derived in convolutional coded parallel combinatorial spread spectrum (PC/SS) systems over additive white Gaussian noise (AWGN) channels. PC/SS systems can achieve higher data transmission than conventional multicode SS systems. To make the derivation tractable, we put a uniform interleaver between a convolutional encoder and a PC/SS transmitter. Since the PC/SS transmitter is employed as the "inner encoder," the bounds are obtained in a similar manner of the derivation in serially concatenated codes through a uniform interleaver. Two different error patterns in the PC/SS system are considered in the performance analysis. Numerical results show that the derived BER bounds are sufficiently accurate. It is found that the coded PC/SS systems outperform coded all-code-parallel DS/SS systems under the same data rate conditions if the number of pre-assigned PN codes increases.