• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.216-222
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• 2016

Generation of a quiet zone in noisy environment is undoubtedly of considerable realistic significance. This paper describes development and implementation of a multichannel real-time active noise control (ANC) system for 3 dimensional noisy environment to enhance the quiet zone performance in terms of size and noise cancellation gain. The proposed ANC system employes a multichannel delay-compensated filtered-X least mean square (FXLMS) algorithm; its real-time implementation is designed in TMS320C6713 digital signal processor (DSP) board. The system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz, and the performance is then illustrated by measuring the quiet zone in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone of quiet with satisfactory size and maximum 24 dB noise attenuation is successfully generated.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.6
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• pp.853-864
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• 2001

In this paper, system model is based on proposed standardization by 3GPP(Asynchronous IMT-2070 system), and we have analyzed the performance of the DS-CDMA/QPSK which cancelled self-interference, so that occurs when the received signal time is delay by multipath fading. We proposed the new scheme that designed for self-interference cancellation and the system performance is calculated in Rician fading and Rayleigh fading channel. The proposed rake receiver can be achieved a gain of about 1 [dB]∼6 [dB] more than generally rake receiver, and will be a very effective method to improve the performance of IMT-2000, and these data can be available for modem design of DS-CDMA.

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

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.13-20
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• 2001

The active noise cancellation system offers a better low frequency performance with a smaller and lighter system compared to a passive one. This paper presents an active noise control system capable of reducing the noise in a helmet after attenuating the external noise using the helmet as the passive noise reduction system, which consists of a controller for inverting and compensating the phase delay, a microphone for picking up the external noise, and a loudspeaker for radiating the acoustic anti-phase signal to reduce the external noise. In this paper, external noise can be reduced by noise controller by compensating the phase difference to be 180°in the frequency of maximun value in the amplitude response. The noise of the phase delay covered from 50°to 310°was reduced in this system and it is possible to obtain a noise reduction of up to approximately 20 dB at the ears in the enclosure.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10C
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• pp.923-929
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• 2003

Turbo processing have been known as methods close to Shannon limit in the aspect of wireless multi-input multi-output (MIMO) communications similarly to wireless single antenna communication. The iterative processing can maximize the mutual effect of coding and interference cancellation, but LDPC coding has not been used for turbo processing because of the inherent decoding process delay. This paper suggests a LDPC coded MIMO system with turbo parallel space-time (Turbo-PAST) processing for high-speed wireless communications and proposes a average soft-output syndrome (ASS) check scheme at low signal to noise ratio (SNR) for the Turbo-PAST system to decide the reliability of decoded frame. Simulation results show that the suggested system outperforms conventional system and the proposed ASS scheme effectively reduces the amount of turbo processing iterations without performance degradation from the point of average number of iterations.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 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 Electronics Engineers of Korea TC
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• v.38 no.12
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• pp.10-19
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• 2001

In this paper, we propose an effective but simple multi-rate parallel interference canceller(PIC) for the international mobile telecommunications-2000(IMT-2000) 3rd generation partnership project (3GPP) system. For effective multi-rate processing, we define the basic block as one symbol period of the dedicated physical control channel(DPCCH) having the lowest data rate and common to all users. Then, decision and interference cancellation are performed at every basic block. For an asynchronous channel, we propose an advance removal scheme that removes in advance multiple access interference(MAI) due to the next blockof other users with shorter delay. Introducing a pipeline structure at a sample base, we can implement efficiently the PIC using the advance removal scheme with a minimum hardware and no extra computations. Through computer simulations, we analyze the bit error rate(BER) performance of the proposed PIC with respect to signal-to-noise ratio(SNR) and the number of users.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

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