• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8A
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• pp.1126-1132
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• 1999

In this paper, we consider the superposed 16-QAM CDMA system for subband coded (SBC) image transmissions. We divide the SBC image bit into two group according to its significance. First, we investigate the average bit error rate (BER) of the overall system when the significance of bits is ignored. Next, we investigate the optimal signal constellation and processing gain that maximizes the peak signal-to-noise ratio (PSNR) of SBC image, subject to a fixed power and bandwidth constraint. We examine the performance improvement over the conventional equal distane, equal processing gain system, and the equal error protection (EEP) system.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.1
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• pp.26-31
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• 2012

The highest frequency band of wavelet decomposition band is divided into 4 equal subbands and by the minimum power of the subbands and by the monotonic transform, the level adapted threshold is obtained. The adapted threshold is applied to the soft threshold technique to denoise high and middle frequency band noise of image signals. And the results of PSNRs are compared with the results obtained by the VisuShrink technique and by the technique using the monotonic transform and the weight value. The results showed the validity of this technique.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.462-470
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• 2002

Wideband speech, characterized by a bandwidth of 50∼7000 ㎐, sounds more natural and intelligible, and is less tiring to listen to when compared to narrowband speech characterized by a bandwidth of 300∼3400 ㎐. Wideband speech coders, however, have not been as successful as the narrowband speech coders because of their higher bit rate. In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. The proposed wideband speech coder operates as follows input speech is first split into two subbands with equal bandwidth and the two subband signals are coded and decoded by each subband coder. A GSM-EFR is adopted as a lower subband coder and a subband coder with wavelet transformed speech is designed for a upper subband coder. The total bit rate of the proposed coder is 18.9kbps (12.2 kbps for lower band coder and 6.7 kbps for upper band coder), and informal listening test results have shown that the proposed coder has comparable speech quality to that of G.722 with 56 kbps.

• Journal of Electrical Engineering and Technology
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• v.11 no.2
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• pp.499-507
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• 2016

Medical imaging techniques such as Magnetic Resonance Imaging (MRI), Computed Tomography (CT) and Ultrasound (US) produce a large amount of digital medical images. Hence, compression of digital images becomes essential and is very much desired in medical applications to solve both storage and transmission problems. But at the same time, an efficient image compression scheme that reduces the size of medical images without sacrificing diagnostic information is required. This paper proposes a novel threshold-based medical image compression algorithm to reduce the size of the medical image without degradation in the diagnostic information. This algorithm discusses a novel type of thresholding to maximize Compression Ratio (CR) without sacrificing diagnostic information. The compression algorithm is designed to get image with high optimum compression efficiency and also with high fidelity, especially for Peak Signal to Noise Ratio (PSNR) greater than or equal to 36 dB. This value of PSNR is chosen because it has been suggested by previous researchers that medical images, if have PSNR from 30 dB to 50 dB, will retain diagnostic information. The compression algorithm utilizes one-level wavelet decomposition with threshold-based coefficient selection.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2001.11b
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• pp.109-113
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• 2001

최근 들어 디지털 이동통신 시스템은 음성 통신 뿐만 아니라 멀티미디어 통신에도 그 적용범위를 확장시키고 있다. 그러나 이러한 시스템에서 한정된 채널 대역폭의 사용은 멀티미디어 정보의 대용량성을 고려할 때, 그 적용범위를 심각하게 축소시키는 가장 큰 제한요소이다. 일반적인 통신 시스템의 소스 부호화기는 채널 잡음을 고려하지 않고 설계되며, 채널 부호화기는 소스 신호의 특성과 무관하게 채널 환경의 극복에만 중점을 두고 설계된다. 그러나 대역폭 제한적인 통신 환경에서 보다 효율적인 대역폭 사용을 위해서는 채널 환경에 따라 소스 부호율과 채널 부호율을 가변적으로 운용하여야 한다. 본 논문에서는 영상을 부대역 부호화하여 각 부대역 영상이 원래의 영상 재구성에 기여하는 중요도에 따라 한정된 채널 자원을 최적으로 할당하는 공동 소스/채널 부호화(Joint Source-channel Coding)에 관하여 연구하였다. 부대역 영상의 소스 부호화로는 DPCM과 PCM을 사용하였고, 채널 부호화는 TCM 부호화기를 사용하였으며 상이오류보호(Unequal Error Protection)를 위해 3가지 채널 부호율에 따라 각각 TCQPSK, TC8PSK, TC16PSK 변조방식을 사용하였다. 모의실험에 사용된 채널 환경은 랜덤잡음 환경과 이동수신의 경우에 발생하는 Rayleigh 페이딩 환경을 고려하였으며, 각 환경에서의 채널 SNR에 따라 동일오류보호(Equal Error Protection) 시스템과 상이오류보호(Unequal Error Protection) 시스템의 성능을 비교하였다.