• 한국통신학회논문지
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• 제33권7B호
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• pp.583-591
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• 2008

본 논문은 다수의 비디오를 전송하는 직교 주파수 다중 분할 시스템의 다운링크에서 종단간 비디오에서 발생하는 왜곡을 최소화하는 자원 할당 방법을 제안한다. 제안된 알고리즘은 제한된 시스템 리소스에서 다수 사용자의 다양성과 패킷 왜곡 모델을 이용한 비트 왜곡 함수를 이용하여 전체 사용자의 비디오 왜곡을 최소화한다. 첫 단계에서, H.264 비디오 코딩 구조에 존재하는 에러 은닉과 전파를 이용하여 비트 왜곡 함수를 유도한다. 두 번째 단계에서는 전체적인 비디오 화질 저하를 최소화하기 위해서, 직교 주파수 다중 분할 시스템의 자원을 사용자의 다양성을 이용하여 부 반송파와 파워를 할당하는 자원 할당 알고리즘을 제안한다. 또한 각 사용자에 의해서 요구되는 비디오 화질을 얻기 위해서, 비례 율의 요소를 적용한다. 실험결과에서, 제안된 자원할당 알고리즘은 기존의 시간 분할 다중 접속 방법 그리고 비트 왜곡의 정보를 이용하지 않는 알고리즘과 비교해서, 종단간의 비디오 화질을 크게 향상시키는 결과를 보였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2005년도 추계종합학술대회
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• pp.323-326
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• 2005

Several video coding standards, such as MPEG-4 and H.263, have been investigated to reduce the resulting number of bits while pursuing the maximum video quality. The recent video coding standard, H.264, provides higher coding efficiency than previous coding standards by using the mode decision scheme. For mode decision, H.264 chooses the best macroblock mode among the several candidates using Lagrangian cost function which reflects both the rate and the distortion. H.264 employs only one rate-distortion optimization (RDO) model for all macroblocks. Since the characteristics of each macroblock is different, each macroblock should have its own RDO model. In this paper, we propose an adaptive rate-distortion optimization algorithm for H.264. We regulate the Lagrangian multiplier considering the picture type and characteristics of each macroblock.

• Journal of Communications and Networks
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• 제7권1호
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• pp.65-75
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• 2005

In this paper, we consider a scenario in which the source has been offered QoS guarantees subject to token-bucket regulation. The rate of the source should be controlled such that it conforms to the token-bucket regulation, and also the distortion obtained is the minimum. We have developed an optimal scheduling algorithm for offline (like pre-recorded video) sources with convex distortion function and which can not tolerate any delay. This optimal offline algorithm has been extended for the real-time online source by predicting the number of packets that the source may send in future. The performance of the online scheduler is not substantially degraded as compared to that of the optimal offline scheduler. A sub-optimal offline algorithm has also been developed to reduce the computational complexity and it is shown to perform very well. We later consider the case where the source can tolerate a fixed amount of delay and derive optimal offline algorithm for such traffic source.

• 음성과학
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• 제10권4호
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• pp.137-147
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• 2003

Speech synthesis can classify by synthesis way, that is waveform coding, source coding and mixture coding. Specially, waveform coding is suitable for high quality synthesis. However, it is not desirable by synthesis techniques of syllable or phoneme unit because it do not separate and handles excitation and formant part. Therefore, there is a need for pitch alteration method applied in synthesis by the rule in waveform coding. This study propose about pitch alteration method that use spectrum scaling after do to flatten spectra by subband linear approximation to minimize spectrum distortion. This paper show evaluation whether show excellency of some measure compared with LPC, Cepstrum, lifter function and method that propose. estimation method seeks distribution of each flattened signal and measured degree of flattened spectra Signal flattened is normalized, So that highest point amounts to zero, and distribution of signal ,whose average is zero, is calculated. this show result that measure the spectrum distortion rate to estimate performance of method that propose. The average spectrum distortion rate was kept below the average 2.12%, so the method that propose is superiors than existent method.

• 한국음향학회지
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• 제21권3호
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• pp.315-322
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• 2002

본 논문에서는 음성 신호 시간축 분할의 새로운 기법으로, 비트율과 왜곡을 함께 고려한 기법이 제안되었다. 시간축 분할에 필요한 보간 함수는 학습 음성 데이터로부터 얻어진다. 보간 함수는 두 타겟간의 길이에 따라 유일하게 결정되므로 보간 함수는 추가 정보없이 표현된다. 타겟 샘플은 비트율을 최소화시키면서 동시에 최대 스펙트럼 오차가 문턱 치보다 작게 되도록 선택하였다. 제안된 기법은 음성 부호화기의 스펙트럼 변수로 널리 사용되는 LSP계수의 부호화에 적용되었으며, 모의실험 결과 평균적으로 8 bits/Frame의 비트율에서 1.4 dB의 스펙트럼 왜곡이 얻어짐을 알 수 있었다.

• 한국소음진동공학회논문집
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• 제13권5호
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• pp.334-340
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• 2003

Exponential window function Is widely used In impact hammer testing to reduce leakage error as well as to get a good S/N ratio. The larger its decaying rate is, the more effectively the leakage errors are reduced. But if the decay rate of the exponential window is too large, the FRF is distorted. And the modal parameters of the system can not be exactly identified by modal analysis technique. Therefore, it is a difficult problem to determine proper decay rate in impact hammer testing. In this paper, amount of the FRF distortion caused by exponential window is theoretically uncovered. A new circle fitting method is also proposed so that the modal parameters are directly extracted from impulse response spectrum distorted by the exponential-windowed impulse response data. The results by the conventional and proposed circle fitting method are compared through a numerical example.

• 전자공학회논문지B
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• 제29B권8호
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• pp.66-73
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• 1992

This paper proposes a new motion estimation algorithm which takes into account the rate-distortion relation in encoding motion compensated error images. The proposed algorithm is based on a new block-matching criterion which is the function of not only the mean squared block-matching error but also the code length for the entropy coded motion vector. The proposed algorithm optimizes the trade-off between the bit rate for motion compensated error images and the bit rate for the motion vectors. Simulation results show that in the motion compensated image coding the proposed motion estimator improves the overall performance by 0.5 dB when compared to the motion estimator which uses MSE only.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(1)
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• pp.233-236
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• 2001

Orthogonal freqency division multiplexing (OFDM) is an attractive technique for achieving high-bit-rate wireless data transmission. However, the potentially large peak-to-average power ratio (PAPR) has limited its application; An OFDM signal with the large PAPR can cause power degradation (In-band distortion) and spectral spreading (Out-of-band distortion) by being clipped passing through a power amplifier. Thus, we propose the combining algorithm of Hadamard transform and phase shift, which is ascribed to the relation between the correlation of the IFFT input sequence function and PAPR. Extensive computer simulations show that the combining algorithm is an effective technique to reduce PAPR.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.291-292
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• 2006

This paper presents rate-distortion optimization that is considered sequence statistics(complexity) to choose the best macroblock mode decision in H.264. In previous work, Lagrange multiplier is derived by the function of constant value 0.85 and QP so that is not the proper Lagrange multilplier for any image sequence. The proposed algorithm solves the problem by changing constant value 0.85 into adaptive value which is influenced by image complexity, and by reducing the encoder complexity to estimate the image statistics with the multiplication of transformed, quantized rate and distortion. Proposed algorithm is achieved the bit-rate saving up to 5% better than previous method.

• 말소리와 음성과학
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• 제3권3호
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• pp.91-98
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• 2011

In this paper, a multi-channel speech enhancement method using the linearly constrained minimum variance (LCMV) algorithm and a variable learning rate control is proposed. To control the learning rate for adaptive filters of the LCMV algorithm, the direction of arrival (DOA) is measured for each short-time input signal and the likelihood function of the target speech presence is estimated to control the filter learning rate. Using the likelihood measure, the learning rate is increased during the pure noise interval and decreased during the target speech interval. To optimize the parameter of the mapping function between the likelihood value and the corresponding learning rate, an exhaustive search is performed using the Bark's scale distortion (BSD) as the performance index. Experimental results show that the proposed algorithm outperforms the conventional LCMV with fixed learning rate in the BSD by around 1.5 dB.