Dependent Quantization (DQ) is a new quantization tool introduced in the Versatile Video Coding (VVC) standard. While it provides better rate-distortion calculation accuracy, it also increases the computational complexity and hardware cost compared to the widely used scalar quantization. To address this issue, this paper proposes a parallel-dependent quantization hardware architecture using Verilog HDL language. The architecture preprocesses the coefficients with a scalar quantizer and a high-frequency filter, and then further segments and processes the coefficients in parallel using the Viterbi algorithm. Additionally, the weight bit width of the rate-distortion calculation is reduced to decrease the quantization cycle and computational complexity. Finally, the final quantization of the TU is determined through sequential scanning and judging of the rate-distortion cost. Experimental results show that the proposed algorithm reduces the quantization cycle by an average of 56.96% compared to VVC’s reference platform VTM, with a Bjøntegaard delta bit rate (BDBR) loss of 1.03% and 1.05% under the Low-delay P and Random Access configurations, respectively. Verification on the AMD FPGA development platform demonstrates that the hardware implementation meets the quantization requirements for 1080P@60Hz video hardware encoding.

The achievability part of the rate-distortion theorem is proved by showing existence of good codes. For i.i.d. sources, two methods showing existence are known; random coding and non-random coding. For general sources, however, no proof in which good codes are constructed with non-random coding is found. In this paper, with a non-random method of code construction, we prove the achievability part of the rate-distortion theorem for general sources. Moreover, we also prove a stochastic variation of the rate-distortion theorem with the same method.

We discuss the spectral efficiency of orthogonal frequency-division multiplexing (OFDM) signals widely adopted in practical systems from a viewpoint of their power spectral density property. Since the conventional OFDM does not make use of pulse shaping filter, its out-of-band (OOB) spectrum may not be negligible especially when the number of subcarriers is small. Thus, in practice, windowing is applied to mitigate OOB emission by smoothing the transition of consecutive OFDM symbols, but its effectiveness has not been well investigated. Furthermore, OFDM signal suffers from nonlinear distortion associated with its high signal peak-to-average power ratio (PAPR), which also leads to OOB radiation. We examine how power amplifier nonlinearity affects the spectral efficiency based on the theoretical results developed in the literature.

Methods that embed data into printed images and retrieve data from printed images captured using the camera of a mobile device have been proposed. Evaluating these methods requires printing and capturing actual embedded images, which is burdensome. In this paper, we propose a method for reducing the workload for evaluating the performance of data embedding algorithms by simulating the degradation caused by printing and capturing images using generative adversarial networks. The proposed method can represent various captured conditions. Experimental results demonstrate that the proposed method achieves the same accuracy as detecting embedded data under actual conditions.

This paper proposes a low-power 100Gb/s four-level pulse amplitude modulation driver (PAM-4 Driver) based on linear distortion compensation structure for thin-film Lithium Niobate (LiNbO3) modulators, which manages to achieve high linearity in the output. The inductive peaking technology and open drain structure enable the overall circuit to achieve a 31-GHz bandwidth. With an area of 0.292 mm2, the proposed PAM-4 driver chip is designed in a 65-nm process to achieve power consumption of 37.7 mW. Post-layout simulation results show that the power efficiency is 0.37 mW/Gb/s, RLM is more than 96%, and the FOM value is 8.84.

Global Navigation Satellite System (GNSS) receivers often realize anti-jamming capabilities by combining array antennas with space-time adaptive processing (STAP). Unfortunately, in suppressing the interference, basic STAP degrades the GNSS signal. For one thing, additional carrier phase errors and code phase errors to the GNSS signal are introduced; for another, the shape of the cross-correlation function (CCF) will be distorted by STAP, introducing tracking errors when the receiver is in tracking mode. Both of them will eventually cause additional Pseudo-Range (PR) bias, and these problems prevent STAP from being directly applied to high-precision satellite navigation receivers. The paper proposes a novel anti-jamming method based on STAP that solves the above problems. First, the proposed method constructs a symmetric STAP by constraining the STAP coefficients. Subsequently, with the information of the steering vector, a compensation FIR filter is cascaded after the symmetric STAP. This approach ensures that the proposed method introduces only a fixed offset to the code phase and carrier phase, and the order of the STAP completely determines the offset, which can be compensated during PR measurements. Meanwhile, the proposed method maintains the symmetry of the CCF, and the receiver can accurately track the carrier phase and code phase in tracking mode. The effectiveness of the proposed method is validated through simulations, which suggest that, in the worst case, our method does not increase carrier and code phase errors and tracking error at the expense of only a 2.86dB drop in interference suppression performance.

This letter investigates the information-theoretic privacy-utility tradeoff. We analyze the minimum information leakage (f-leakage) under the utility constraint that the excess distortion probability is allowed up to ε∈[0, 1). The derived upper bound is characterized by the ε-cutoff random transformation and a distortion ball.

Since HEVC intra rate control has no prior information to rely on for coding, it is a difficult work to obtain the optimal λ for every coding tree unit (CTU). In this paper, a convolutional neural network (CNN) based intra rate control is proposed. Firstly, a CNN with two last output channels is used to predict the key parameters of the CTU R-λ curve. For well training the CNN, a combining loss function is built and the balance factor γ is explored to achieve the minimum loss result. Secondly, the initial CTU λ can be calculated by the predicted results of the CNN and the allocated bit per pixel (bpp). According to the rate distortion optimization (RDO) of a frame, a spatial equation is derived between the CTU λ and the frame λ. Lastly, The CTU clipping function is used to obtain the optimal CTU λ for the intra rate control. The experimental results show that the proposed algorithm improves the intra rate control performance significantly with a good rate control accuracy.

For improving the fifth-generation mobile communication system, a highly efficient power amplifier must be designed for the base station. An outphasing amplifier is expected to be a solution for achieving high efficiency. We designed a combiner, one of the key components of the outphasing amplifier, using a serial Chireix combiner and fabricated an amplifier with a GaN HEMT, achieving 70% or more high efficiency up to 9 dB back-off power in an 800 MHz band. We also fabricated a 2 GHz-band outphasing amplifier with the same design. We applied digital predistortion (DPD) to control the balance of amplifying units in this amplifier and achieved an average efficiency of 65% under a 20 MHz modulation bandwidth.

This paper presents linearization technologies for high efficiency power amplifiers of cellular base stations. These technologies are important to actualizing highly efficient power amplifiers that reduce power consumption of the base station equipment and to achieving a sufficient non-linear distortion compensation level. It is well known that it is very difficult for a power amplifier using linearization technologies to achieve simultaneously high efficiency and a sufficient non-linear distortion compensation level. This paper presents two approaches toward addressing this technical issue. The first approach is a feed-forward power amplifier using the Doherty amplifier as the main amplifier. The second approach is a digital predistortion linearizer that compensates for frequency dependent intermodulation distortion components. Experimental results validate these approaches as effective for providing power amplification for base stations.

This paper presents an efficient method for solving PnP, PnPf, and PnPfr problems, which are the problems of determining camera parameters from 2D-3D point correspondences. The proposed method is derived based on a simple usage of linear algebra, similarly to the classical DLT methods. Therefore, the new method is easier to understand, easier to implement, and several times faster than the state-of-the-art methods using Gröbner basis. Contrary to the existing Gröbner basis methods, the proposed method consists of three algorithms depending on the number of the points and the 3D point configuration. Experimental results show that the proposed method is as accurate as the state-of-the-art methods even in near-planar scenes while achieving up to three times faster.

Orthogonal frequency division multiplexing (OFDM) technique has been widely used in communication systems in pursuit of the most efficient utilization of spectrum. However, the increase of the number of orthogonal subcarriers will lead to the rise of the peak-to-average power ratio (PAPR) of the waveform, thus reducing the efficiency of the power amplifiers. In this letter we propose a phase-changed PAPR reduction technique based on windowing function architecture for OFDM systems. This technique is based on the idea of phase change, which makes the spectrum of output signal almost free of regrowth caused by peak clipping. It can reduce more than 28dBc adjacent channel power ratio (ACPR) compared with the traditional peak windowing clipping methods in situation that peak is maximally suppressed. This technique also has low algorithm complexity so it can be easily laid out on hardware. The proposed algorithm has been laid out on a low-cost field-programmable gate array (FPGA) to verify its effectiveness and feasibility. A 64-QAM modulated 20M LTE-A waveform is used for measurement, which has a sampling rate of 245.67M.

A new structure of the electro-optic modulator to compensate the third-order intermodulation distortion (IMD3) is introduced. The modulator includes two Mach-Zehnder modulators (MZMs) operating with frequency chirp and the two modulated outputs are combined with an adequate phase difference. We revealed by theoretical analysis and numerical calculations that the IMD3 components in the receiver output could be selectively suppressed when the two MZMs operate with chirp parameters of opposite signs to each other. Spectral power of the IMD3 components in the proposed modulator was more than 15dB lower than that in a normal Mach-Zehnder modulator at modulation index between 0.15π and 0.25π rad. The IMD3 compensation properties of the proposed modulator was experimentally confirmed by using a dual parallel Mach-Zehnder modulator (DPMZM) structure. We designed and fabricated the modulator with the single-chip structure and the single-input operation by integrating with 180° hybrid coupler on the modulator substrate. Modulation signals were applied to each modulation electrode by the 180° hybrid coupler to set the chirp parameters of two MZMs of the DPMZM. The properties of the fabricated modulator were measured by using 10GHz two-tone signals. The performance of the IMD3 compensation agreed with that in the calculation. It was confirmed that the IMD3 compensation could be realized even by the fabricated modulator structure.

This paper presents analysis results of the intermodulation distortion (IMD) components compensation conditions for dual-band feed-forward power amplifier (FFPA) when inputting dual-band signals simultaneously. The signal cancellation loop and distortion cancellation loop of the dual-band FFPA have frequency selective adjustment paths which consist of filter and vector regulator. The filter selects the desired frequency component and suppresses the undesired frequency component in the desired frequency selective adjustment path. The vector regulators repeatedly adjust the amplitude and phase values of the composite components for the desired and suppressed undesired frequency components. In this configuration, the cancellation levels of the signal cancellation loop and distortion cancellation loop are depending on the amplitude and phase errors of the vector regulator. The analysis results show that the amplitude and phase errors of the desired frequency component almost become independent that of the undesired frequency component in a weak non-linearity condition, when the isolation between the desired band and the undesired band given by the filter is more than 40 dB. The amplitude errors of the desired frequency component are dependent on that of the undesired frequency component in a strong non-linear conditions when the isolation level sets as above. A 1-W-class signal cancellation loop and 20-W-class FFPA are fabricated for 1.7-GHz and 2.1-GHz bands simultaneous operation. The experimental results show that the analysis results are suitable in the experimental conditions. From these investigations, the analysis results can provide a commercially available dual-band FFPA. To our best knowledge, this is first analysis results for the dual-band FFPA.

Broadband amplifiers have been used in modern wireless communication systems. However, the accompanying disadvantage is that there is more nonlinear interference in the available operating frequency band. In addition to the in-band intermodulation distortion which affecting adjacent frequency bands the most important is harmonic distortion. In this letter we present a robust and low complex digital harmonic canceling model called cross-disturbing harmonic (CDH) model for broadband power amplifiers (PAs). The approach introducing cross terms is used to enhance the robustness of the model, thereby significantly increase the stability of the system. The CDH model still has excellent performance when actively reducing the number of coefficients. Comparisons are conducted between the CDH model and the other state-of-the-art model called memory polynomial harmonic (MPM) model. Experimental results show that the CDH model can achieve comparable performance as the MPM model but with much fewer (43%) coefficients.

In this article, we propose a method called “continuous noise masking (cNM)” that allows eliminating residual buzziness in a continuous vocoder, i.e. of which all parameters are continuous and offers a simple and flexible speech analysis and synthesis system. Traditional parametric vocoders generally show a perceptible deterioration in the quality of the synthesized speech due to different processing algorithms. Furthermore, an inaccurate noise resynthesis (e.g. in breathiness or hoarseness) is also considered to be one of the main underlying causes of performance degradation, leading to noisy transients and temporal discontinuity in the synthesized speech. To overcome these issues, a new cNM is developed based on the phase distortion deviation in order to reduce the perceptual effect of the residual noise, allowing a proper reconstruction of noise characteristics, and model better the creaky voice segments that may happen in natural speech. To this end, the cNM is designed to keep only voice components under a condition of the cNM threshold while discarding others. We evaluate the proposed approach and compare with state-of-the-art vocoders using objective and subjective listening tests. Experimental results show that the proposed method can reduce the effect of residual noise and can reach the quality of other sophisticated approaches like STRAIGHT and log domain pulse model (PML).

Adaptive Volterra filters (AVFs) are usually used to identify nonlinear systems, such as loudspeaker systems, and ordinary adaptive algorithms can be used to update the filter coefficients of AVFs. However, AVFs require huge computational complexity even if the order of the AVF is constrained to the second order. Improving calculation efficiency is therefore an important issue for the real-time implementation of AVFs. In this paper, we propose a novel sub-band AVF with high calculation efficiency for second-order AVFs. The proposed sub-band AVF consists of four parts: input signal transformation for a single sub-band AVF, tap length determination to improve calculation efficiency, switching the number of sub-bands while maintaining the estimation accuracy, and an automatic search for an appropriate number of sub-bands. The proposed sub-band AVF can improve calculation efficiency for which the dominant nonlinear components are concentrated in any frequency band, such as loudspeakers. A simulation result demonstrates that the proposed sub-band AVF can realize higher estimation accuracy than conventional efficient AVFs.

It is generally believed that jamming signals similar to communication signals tend to demonstrate better jamming effects. We believe that the above conclusion only works in certain situations. To select the correct jamming scheme for a multi-level quadrature amplitude modulation (MQAM) signal in a complex environment, an optimal jamming method based on orthogonal decomposition (OD) is proposed. The method solves the jamming problem from the perspective of the in-phase dimension and quadrature dimension and exhibits a better jamming effect than normal methods. The method can construct various unconventional jamming schemes to cope with a complex environment and verify the existing jamming schemes. The Experimental results demonstrate that when the jammer ideally knows the received power at the receiver, the proposed method will always have the optimal jamming effects, and the constructed unconventional jamming scheme has an excellent jamming effect compared with normal schemes in the case of a constellation distortion.

Design of an equivalent slew-rate monolithic pulse generator using bipolar-CMOS-DMOS (BCD) technology for medical ultrasound transmitters is presented in this paper. The pulse generator employs a floating capacitive coupling level-shifter architecture to produce a high-voltage (Vpp=80V) output. The performance of equivalent slew-rate in the rising and falling edge is achieved by carefully choosing the value of coupling capacitors and the size of the final stage high-voltage MOSFETs of the pulse generator. The measured output pulses show the rising and falling time of 8.6nsec and 8.5nsec, respectively with second harmonic distortion down to -40dBc, indicating the designed pulse generator can be used for advanced ultrasonic harmonic imaging systems.

We have proposed a novel concept of a digital watermarking technique for music data that focuses on the use of sound synthesis and sound effect techniques. This paper describes the details of our proposed technique that employs the distortion effect, one of the most common sound effects frequently utilized especially for guitar and bass instruments. This paper describes the experimental results of evaluating the resistance of the proposed technique against some basic malicious attacks utilizing MP3 coding, tempo alteration, pitch alteration, and high-pass filtering. It is demonstrated that the proposed technique potentially has appropriate resistance against such attacks except for the high-pass filtering attack. A technique for increasing the resistance against the high-pass filtering attack is also supplementarily discussed.