• The Transactions of The Korean Institute of Electrical Engineers
• /
• v.64 no.12
• /
• pp.1756-1760
• /
• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.474-476
• /
• 2009

In the radio communication, such as echo cancellation and channel equalization, adaptive filtering is very practical. Its convergence behavior that is used for updating the weights depends on the correlation of the input signal and length of adaptive filter. Highly correlated input and long length of adaptive filter deteriorate the convergence behavior. To solve this problem, recently, subband affine projection algorithm which pre-whiten the correlation of the input and update the weights in subband structure has been presented. This paper presents convergence analysis method of multiple constrained subband affine projection algorithm.

• ETRI Journal
• /
• v.45 no.5
• /
• pp.874-886
• /
• 2023

A joint resource-optimization scheme is investigated for nonorthogonal multiple access (NOMA)-enhanced scalable video coding (SVC) multicast in unmanned aerial vehicle (UAV)-assisted radio-access networks (RANs). This scheme allows a ground base station and UAVs to simultaneously multicast successive video layers in SVC with successive interference cancellation in NOMA. A video quality-maximization problem is formulated as a mixed-integer nonlinear programming problem to determine the UAV deployment and association, RAN spectrum allocation for multicast groups, and UAV transmit power. The optimization problem is decoupled into the UAV deployment-association, spectrum-partition, and UAV transmit-power-control subproblems. A heuristic strategy is designed to determine the UAV deployment and association patterns. An upgraded knapsack algorithm is developed to solve spectrum partition, followed by fast UAV power fine-tuning to further boost the performance. The simulation results confirm that the proposed scheme improves the average peak signal-to-noise ratio, aggregate videoreception rate, and spectrum utilization over various baselines.

• ETRI Journal
• /
• v.46 no.3
• /
• pp.446-460
• /
• 2024

In this paper, we mathematically investigate a downlink non-orthogonal multiple access (NOMA) system for short-packet communications (SPC) in which the near users are used as full-duplex (FD) relays to forward intended signals from the source to a far user. In addition, partial relay selection is employed to enhance the performance of the FD relays under the impact of imperfect interference cancellation. At the far user, selection combining (SC) or maximal ratio combining (MRC) is employed to combine the signals received from the source and the selected FD relay. The analytical expressions for the average block error rate (BLER) of two users over flat Rayleigh fading channels are derived. Furthermore, closed-form asymptotic expressions of the average BLERs at the near and far users in high signal-to-noise ratio (SNR) regimes are obtained. The numerical results show that the analytical BLERs of the near user and far user closely match the simulation results.

• International Journal of Fuzzy Logic and Intelligent Systems
• /
• v.7 no.4
• /
• pp.228-235
• /
• 2007

The main purpose of this paper is to manage the container property effectively at the container yard by applying the RTLS technology to the field of port logistics. Yet, many kinds of noises happen to be inputted with the distance value(between the reader and the tag) which is to be inputted into the location identification algorithm, which makes the distance value jumped due to the system noise of the ultrasonic sensor module and the measurement noise. The Kalman Filter is widely used to prevent this jump occurrence; the noises are eliminated by using the EKF(Extended Kalman Filter) while considering that the distance information of the ultrasonic sensor is non-linear. Also, the 3D RTLS system at the port container yard suggested in this research is designed not to be interrupted for its ultrasonic transmission by positioning the antenna at the front of each sector of the container where the active tags are installed. We positioned the readers, which function as antennas for location identification, to four places randomly in the absolute coordinate and let the positions of the active tags identified by using the distance data delivered from the active tags. For the location identification algorithm used in this paper, the triangulation measurement that is most used in general is applied and newly reorganized to calculate the position of the container. In the first experiment, we dealt with the error resulting in the angle and the distance of the ultrasonic sensor module, which is the most important in the hardware performance; in the second, we evaluated the performance of the location identification algorithm, which is the most important in the software performance, and tested the noise cancellation effects for the EKF. According to the experiment result, the ultrasonic sensor showed an average of 3 to 5cm error up to $45^{\circ}$ in case of $60^{\circ}$ or more, non-reliable linear distances were obtained. In addition, the evaluation of the algorithm performance showed an average of $4^{\circ}{\sim}5^{\circ}$ error due to the error of the linear distance-this error is negligible for most container location identifications. Lastly, the experiment results of noise cancellation and jump preservation by using the EKF showed that noises were removed in the distance information which was entered from the input of the ultrasonic sensor and as a result, only signal was extracted; thus, jumps were able to be removed and the exact distance information between the ultrasonic sensors could be obtained.

• Korean Journal of Optics and Photonics
• /
• v.21 no.3
• /
• pp.103-110
• /
• 2010

A fiber-phase-noise compensating system was constructed for a 1.4 GHz reference frequency transferred through a 13-km-long fiber spool. The transfer instability was dependent on the temperature variation of the compensating system. With the room temperature variation stabilized within $0.3^{\circ}C$, the transfer instability was $4.6{\times}10^{-14}$ at 0.8 s of average time and $2.5{\times}10^{-16}$ at 1000 s of average time with the fiber phase noise compensated. However, with the room temperature changed by $3.5^{\circ}C$, the transfer instability was $6.8{\times}10^{-14}$ at 1.2 s of average time and $3.0{\times}10^{-15}$ at 1000 s of average time. From this result, the temperature stability condition for the experimental setup could be determined to obtain a transfer instability of $10^{-16}$ at 1000 s of average time.

• The Journal of the Acoustical Society of Korea
• /
• v.29 no.7
• /
• pp.458-467
• /
• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.

• Journal of the Institute of Convergence Signal Processing
• /
• v.14 no.2
• /
• pp.110-116
• /
• 2013

The amplification rate of a RF repeater is limited by the feedbacked signals from the same repeater. And an ICS (Interference Cancellation System) repeater has been developed to remove the feedbacked signals. The ICS repeater estimates the amplitudes and the phases of the feedbacked signals and removes the estimated feedback signals from the received input signal of the repeater. However, it requires lots of hardware complexity and this leads to the increase the cost of the repeater. Moreover, the ICS repeater can not solve the pilot pollution problems. To solve these problems, we have studied the implementation and adaptation of smart antenna system for RF repeaters. We have designed a smart antenna system with a switching beam structure in order to reduce the hardware and computational complexity. After analyzing the proposed smart antenna system, we found out that the amplification rate of the proposed repeater increases 23dB compare to the amplification rate of ICS repeater and the output SINR increases 6dB compare to the ICS repeater.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.18 no.7
• /
• pp.839-846
• /
• 2007

In this paper, the first attempt to design a novel structure of dual-band feedforward linear power amplifier(FFW LPA) was presented. Up to now, primary technical difficulty has been the extension of the conventional signal canceller to the dual-band operation. Therefore, we propose the design technique of the dual-band equal group delayed carrier canceller, the dual-band equal group delayed intermodulation distortion(IMD) canceller and the dual-band FFW LPA. The operation frequency bands of the implemented dual-band FFW LPA are digital cellular($f_0=880$ MHz) and IMT-2000($f_0=2.14$ GHz) band, which are separated about 1.26 GHz. With the high power amplifier of 120 W PEP for commercial base-station application, IMD cancellation loop shows 20.45 dB and 25.04 dB loop suppression at each band of operation for 100 MHz. From the adjacent channel leakage ratio(ACLR) measurement with CDMA IS-95A 4FA and WCDMA 4FA signal, we obtained 16.52 dB improvement at the average output power of 41.5 dBm for digital cellular band, and 18.59 dB improvement at the average output power of 40 dBm for IMT-2000 band simultaneously.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.2
• /
• pp.68-77
• /
• 2005

The purpose of a spatial audio system is to give a listener an impression as if he were present in a recorded environment when its sound is reproduced. For this purpose a dummy head microphone is generally used. Because of its human-like shape, dummy head microphone can reproduce spatial images through headphone reproduction. However, its shape and size are restriction to public use and it is difficult to convert the output signal of dummy head microphone into a multi-channel signal for multi-channel environment. So, in this paper, we propose a multi-channel 3D microphone technology. The multi-channel 3D microphone acquire a spatial audio using five microphones around a horizontal plane of a rigid sphere and through post processing, it can reproduce various reproduction signals for headphone, stereo, stereo dipole, 4ch and 5ch reproduction environments. Because of complex computation, we implemented H/W based post processing system. To verily the Performance of the multi-channel 3D microphone, localization experiments were Performed. The result shows that a front/back confusion, which is the one of common limitations of conventional dummy head technology, can be reduced dramatically.