• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2002.11a
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• pp.229-234
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• 2002

OFDM(Orthogonal Frequency Division Multiplexing) 전송방식은 SCM(Single Carrier Modulation)에 비해 우수한 여러 가지 장점들을 가지며, 방송시스템들 중 Eureka 147 DAB(Digital Audio Broadcasting) 시스템에 가장 먼저 채택되었다. 그러나 OFDM 신호의 높은 PAPR(Peak-to-Average Power Ratio) 특성은 D/A, A/D 변환기의 복잡도를 높이고, 고출력 증폭기의 효율성을 감소시키는 원인이 된다. 이를 개선하기 위한 방법 중에, SDT(Signal Distortion Technique)는 전송시스템의 규격 및 수신기의 변경 없이도 적용 가능하다는 장점을 갖는다. 본 논문에서는 SDT에 속하는 companding 기법을 Eureka 147 DAB 시스템에 적용하여 PAPR 개선정도에 따른 시스템의 요구 $E_2/N_0$ 및 out-of-band의 PSD 열화 정도를 분석하였으며, 이를 clipping 기법의 성능과 비교하였다. 모의실험 결과, $\mu$값이 2인 경우, companding 기법이 PAPR, $E_2/N_0$, out-of-band의 PSD 특성 모두에서 clipping 기법에 비해 우수한 성능을 나타냈다. 또한 $\mu$ 값을 고정시킨 경우, 정규화 값이 증가할수록 신호왜곡 정도가 줄어들어 $E_2/N_0$, out-of-band의 PSD 성능개선 정도는 증가하지만, 이와는 반대로 PAPR 값은 개선 정도가 줄어들었다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.15 no.6
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• pp.1-7
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• 1978

This paper deals with speech digitization and bandwidth compression techniques, particularly two predictive coding methods-namely, adaptive differential pulse code modulation(ADPCM) and adaptive delta modulation(ADM). The principle of a typical adaptive quantizer that is used in ADPCM is explained, and discussed. Also, three companding methods(instantaueous, syllabic, and hybrid companding) that are used in ADM are explained in detail, and their performances are compared. In addition, the performances of ADPCM and ADM as speech coders are compared, and the inerits of each coder are discussed.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2022.06a
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• pp.61-62
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• 2022

본 논문에서는 시간 영역 에너지의 companding을 이용하여 오디오 부호화에서 발생하는 프리 에코를 효과적으로 감소시키는 방법을 제안한다. 일반적으로 오디오 부호화는 블록 단위의 변환 부호화를 사용하므로 과도 구간에서 프리 에코를 발생시킨다. 프리 에코를 줄이기 위한 기존 TNS 방법은 주파수 영역에서 선형 예측 방법을 사용하며, 부가 정보 전송이 필요하고 성능이 낮은 문제점을 가진다. 제안하는 방법은 시간 영역 에너지의 동적 범위를 감소시킨 후 부호화 하고, 복호화 이후에 에너지를 복원하는 과정을 통하여 양자화 오차의 시간 영역 에너지 분포를 조정하여 프리 에코를 감소시킨다. 제안하는 방법이 TNS보다 우수한 프리 에코 감소 성능을 가지는 것을 확인하였다.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1909-1912
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• 2002

We propose in this paper a companding scheme or peak-to-average power ratio (PAPR) reduction in orthogonal frequency division multiplexing (OFDM) systems. By exploiting statistical distribution of OFDM transmit signals, the proposed scheme effectively reduces the PAPR by compressing the peak signals, while maintaining the average power unchanged. Simulation results are provided to show good performance of the proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.2
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• pp.202-206
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• 1993

To secure the 64 kbps clear channel capability in the ISDN, it was decided to convert its North American digital transmission system into the European system especially in DS1 level. But their different companding law became an issue in the process of R&D in the existing time division digital switch such as TDX-1 to incorporate it into the ISDN. TDX-1 has been installed with North American ${\mu}$ companding law and its ISDN capability is under development now. Thus, because of the ${\mu}$ law/A law conversion for the interworking of two different transmission system, it is difficult to maintain the ISDN B channel transparency between TDX-1 and new European system with A companding law. Among some solutions for it, European frame format with ${\mu}$ law companding is recommended. Those problems and solutions are presented in this paper.

• The Journal of the Acoustical Society of Korea
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• v.10 no.2
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• pp.44-51
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• 1991

In this paper, we have studied the FM-extended broadcasting which makes the enhancement of stereo signal-to-noise ration by utilizing the compression technique and quadrature moduation method of standard subcarrier signal in transmitter station, and corresponding demodulation method and expansion technique in receiver system. The proposed system is completely compatible with the conventional FM stereo system. We have confirmed the validity of the proposed method by implementation of transmitter/receiver system including quadrature modulation/demodulation and companding system.

• Journal of the Korean Institute of Telematics and Electronics
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• v.15 no.3
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• pp.1-6
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• 1978

This paper deals with speech digitization and bandwidth compression techniques, particularly two predictive coding methods-namely, adaptive diferentia1 pulse code modulation(ADPCM) and adaptive delta modulation (ADM). The principle of a typical adoptive quantizer that is used in ADPCM is explained, and two analysis methods for the adaptive predictor coefficents, block and sequential analyses, are discussed. Also, three companding methods (instantaneous, syllabic, and hybrid companding) that are used in ADM are explained in detail, and their performances are compared. In addition, the performances of ADPCM and ADM as speech coders are compared, and the merits of each coder are discussed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.5 no.1
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• pp.54-59
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• 1980

Adaptive version of the delta modulator that is akin to wyllabic companding for telephony is described. A digital technique is used to sense the slope of the input signal and to control the amplitude of the pulses supplied to the RC network in the feed back loop. Thus the development was stimulated by the suitability of delta modulator for low-cost integrated circuits. Analysis are made of the optimum overload characteristics, stability, SNR for improving the frequency range and these results have been esperimentally verfied.

• ETRI Journal
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• v.43 no.2
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• pp.197-208
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• 2021

Single-carrier frequency division multiple access (SC-FDMA) is currently being used in long-term evolution uplink communications owing to its low peak-to-average power ratio (PAPR). This study proposes a new transceiver design for an SC-FDMA system based on Walsh-Hadamard transform (WHT). The proposed WHT-based SC-FDMA system has low-PAPR and better bit-error rate (BER) performance compared with the conventional SC-FDMA system. The WHT-based SC-FDMA transmitter has the same complexity as that of discrete Fourier transform (DFT)-based transmitter, while the receiver's complexity is higher than that of the DFT-based receiver. The exponential companding technique is used to reduce its PAPR without degrading its BER. Moreover, the performances of different ordered WHT systems have been studied in additive white Gaussian noise and multipath fading environments. The proposed system has been verified experimentally by considering a real-time channel with the help of wireless open-access research platform hardware. The supremacy of the proposed transceiver is demonstrated based on simulated and experimental results.

• Journal of Biomedical Engineering Research
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• v.28 no.6
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• pp.732-743
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• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.