• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.583-591
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• 2008

This paper proposes a resource allocation algorithm for minimizing the overall distortion of multiple users in orthogonal frequency division multiplex (OFDM). The proposed algorithm exploits the diversity of multiuser and the rate-distortion function using packet distortion model in a system with limited resources. We first induce a rate-distortion function considering error concealment and error propagation properties of H.264 video structures. Then we perform adaptive resource allocation utilizing multiuser diversity for minimizing the overall video quality degradation. We also consider the proportional rate which is pre-determined for each user. Simulation results show that compared to the previous time division multiple access method and the resource allocation method maximizing data rate, the proposed rate allocation algorithm substantially improves the received video quality.

• Proceedings of the IEEK Conference
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• 2005.11a
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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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• v.7 no.1
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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.

• Speech Sciences
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• v.10 no.4
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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.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.315-322
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• 2002

In this paper, a new temporal decomposition method is proposed. which takes into consideration not only spectral distortion but also bit rates. The interpolation functions, which are one of necessary parameters for temporal decomposition, are obtained from the training speech corpus. Since the interval between the two targets uniquely defines the interpolation function, the interpolation can be represented without additional information. The locations of the targets are determined by minimizing the bit rates while the maximum spectral distortion maintains below a given threshold. The proposed method has been applied to compressing the LSP coefficients which are widely used as a spectral parameter. The results of the simulation show that an average spectral distortion of about 1.4 dB can be achieved at an average bit rate of about 8 bits/Frame.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.13 no.5
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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.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.29B no.8
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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.

• Proceedings of the IEEK Conference
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• 2001.06a
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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.

• Proceedings of the IEEK Conference
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• 2006.06a
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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.

• Phonetics and Speech Sciences
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• v.3 no.3
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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.