• Journal of Broadcast Engineering
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• v.21 no.6
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• pp.929-943
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• 2016

This paper proposes the rate control algorithm based on the S-JND (Saliency-Just Noticeable Difference) model for considering perceptual visual quality. The proposed rate control algorithm employs the S-JND model to simultaneously reflect human visual sensitivity and human visual attention for considering characteristics of human visual system. During allocating bits for CTU (Coding Tree Unit) level in a rate control, the bit allocation model calculates the S-JND threshold of each CTU in a picture. The threshold of each CTU is used for adaptively allocating a proper number of bits; thus, the proposed bit allocation model can improve perceptual visual quality. For performance evaluation of the proposed algorithm, the proposed algorithm was implemented on HM 16.9 and tested for sequences in Class B and Class C under the CTC (Common Test Condition) RA (Random Access), Low-delay B and Low-delay P case. Experimental results show that the proposed method reduces the bit-rate of 2.3%, and improves BD-PSNR of 0.07dB and bit-rate accuracy of 0.06% on average. We achieved MOS improvement of 0.03 with the proposed method, compared with the conventional method based on DSCQS (Double Stimulus Continuous Quality Scale).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.614-622
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• 2006

In this paper, we propose an efficient packetization method to reduce the packetization overhead. For the purpose, we firstly verify the relationship between packet length and packet loss rate. The empirical results show that as the packet length is larger than the path MTU, the packet loss rate is drastically increased, producing poor visual quality at the receiver side. However, as the length of the packet is reduced, we should transmit more packets per frame and the packetization overhead will be increased. This increase in the packetization overhead reduces the number of bits allocated to the video data, resulting in the low visual quality. Therefore, each packet should be packetized to have the packet length close to the path MTU. In this paper, we show that the this process of the packetization with the constraint on the packet length is very similar to the dynamic storage allocation in the operating system. We had thoroughly surveyed the dynamic storage allocation methods used in the recent operating systems and propose to use the allocation methods for the video packetization. We empirically show that the proposed method can reduce the packetization overhead upto 28.3%, compared with the conventional sequential packetization method which have been widely used in Internet video transmission.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.68-77
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• 2000

Line special frequency (LSF) parameters have been widely used in low bit-rate speech coding due to their efficiency for representing the short-time speech spectrum. In this paper, a new distance measure based on the masking properties of human ear is proposed for quantizing LSF parameters whereas most conventional quantization methods are based on the weighted Euclidean distance measure. The proposed method derives the perceptual distance measure from the definition of noise-to-mask ratio (NMR) which has high correspondence with the actual distortion received in the human ear and uses it for quantizing LSF parameters. In addition, we propose an adaptive bit allocation scheme, which allocates minimal bits to LSF parameters maintaining the perceptual transparency of given speech frame for reducing the average bit-rates. For the performance evaluation, we has shown the ratio of perceptually transparent frames and the corresponding average bit-rates for the conventional and proposed methods. By jointly combining the proposed distance measure and adaptive bit allocation scheme, the proposed system requires only 770 bps for obtaining 95.5% perceptually transparent frames, while the conventional systems produce 89.9% at even 1800 bps.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.3
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• pp.69-74
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• 2009

A novel protocol is proposed to achieve sub-carner-based rate adaptation in OFDM-based wireless systems. The protocol requires the addition of one OFDM symbol to the Clear-to-Send (CTS) packet defined in the IEEE 802.11 standard_ When receiving a Ready-To-Send (RTS) packet, the receiver determines the number of bits to be allocated in each sub-carrier through channel estimation. This decision is delivered to the sender using an additional OFDM symbol. That is, bit-allocation over sub-carriers is achieved using only one additional OFDM symbol. The protocol also provides an error recovery process to synchronize the bit-allocation information between the sender and receiver. The protocol enhances the channel efficiency in spite of the overhead of one additional OFDM symbol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.3
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• pp.221-231
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• 1990

In this paper, A new 2-D DCT algorithm is proposed to reduce the computational amount of transform operation using the distribution of the motion compensated error signal and the bit allocation table. In the this algorithm, 2-D Walsh-Hadamard transform is directly computed and then multiplied by a constant matrix. Multiplications are concentrated on the final stage in thie algorithm, thus the computational amount is reduced in proportion to the number of transform coefficients that are excluded from quatization. The computational amount in computing only the DCT coefficients allocated to the bit allocation table is calculated. As the result, the number of multiplications is less thn the algorithm known to have the fewest number of computations when less than 0.6 bits per pixel are allocated to tranform coding for the motion compensated error image in the case of the proposed algorithm. Thus, it shows that the proposed algorithm is valid in reducing the computational loads of transform coding.

• Journal of the Korea Society of Computer and Information
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• v.19 no.12
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• pp.111-121
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• 2014

In this paper, we propose tree-based hybrid query tree architecture utilizing time slot. 4-Bit Pattern Slot Allocation(4-SL) has a 8-ary tree structure and when tag ID responses according to query of the reader, it applies a digital coding method, the Manchester code, in order to extract the location and the number of collided bits. Also, this algorithm can recognize multiple Tags by single query using 4 fixed time slots. The architecture allows the reader to identify 8 tags at the same time by responding 4 time slots utilizing the first bit($[prefix+1]^{th}$, F ${\in}$ {'0' or '1'}) and bit pattern from second ~ third bits($[prefix+2]^{th}{\sim}[prefix+3]^{th}$, $B_2{\in}$ {"00" or "11"}, $B_1{\in}$ {"01" or "10"}) in tag ID. we analyze worst case of the number of query nodes(prefix) in algorithm to extract delay time for recognizing multiple tags. The identification delay time of the proposed algorithm was based on the number of query-responses and query bits, and was calculated by each algorithm.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.

• Journal of Communications and Networks
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• v.16 no.1
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• pp.36-44
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• 2014

Exploiting the energy-delay tradeoff for energy saving is critical for developing green wireless communication systems. In this paper, we investigate the delay-constrained energy-efficient packet transmission. We aim to minimize the energy consumption of multiple randomly arrived packets in an additive white Gaussian noise channel subject to individual packet delay constraints, by taking into account the practical on-off circuit power consumption at the transmitter. First, we consider the offline case, by assuming that the full packet arrival information is known a priori at the transmitter, and formulate the energy minimization problem as a non-convex optimization problem. By exploiting the specific problem structure, we propose an efficient scheduling algorithm to obtain the globally optimal solution. It is shown that the optimal solution consists of two types of scheduling intervals, namely "selected-off" and "always-on" intervals, which correspond to bits-per-joule energy efficiency maximization and "lazy scheduling" rate allocation, respectively. Next, we consider the practical online case where only causal packet arrival information is available. Inspired by the optimal offline solution, we propose a new online scheme. It is shown by simulations that the proposed online scheme has a comparable performance with the optimal offline one and outperforms the design without considering on-off circuit power as well as the other heuristically designed online schemes.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.34-42
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• 2008

MAC(Medium Access Control) is demanded to provide end-to-end QoS(Quality of Service) for a variety of traffic in the wireless networks. When all the traffic is integrated in the channel, the main difficulty of the MAC protocol is how to efficiently support multi-class traffic in the limited bandwidth wireless channel. In this paper, we proposed the dynamic bandwidth slot method for improving QoS of the real time traffics. In this paper, we used in-band scheme to send dynamic parameter and considering buffer size and delay variation, we enabled 2 state bits to send to base station in mobile station. The proposed algorithm is to guarantee QoS of real time traffic and maximize transfer efficiency in wireless networks.