• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.9
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• pp.734-740
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• 2012

In this paper, we propose a method for throughput improvement in multirate multihop wireless networks with intra-session network coding. A multirate transmission supports several rates so that it has a tradeoff between transmission rate and transmission range, therefore multirate can cope with the performance degradation. We present the rate selection method with intra-session network coding in multirate nodes. The effective transmission rate is calculated by using intra-session network coding, and then its maximum effective transmission rate is selected. Concurrent transmission set is determined by selected transmission rate and then the linear programming is formulated for throughput calculation in multihop wireless networks. We evaluate the performance by using MATLAB and lp_solve programs in two topologies. It is shown that the throughput of the proposed method is improved.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.5
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• pp.21-26
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• 2012

Intra-session network coding has been proposed to improve throughput by simplifying scheduling of multi-hop wireless network and efficiency of packet transmission. Multi-rate transmission has been used in multihop wireless networks. An opportunistic routing with multirate shows throughput improvement compared with single rate. In this paper, we propose a method of throughput improvement in multi-hop wireless network by using multi-rate and intra-session network coding. We suggest a method to select an local optimal transmission rate at each node. The maximum throughput is evaluated by using linear programming (LP). To solve the LP, we use MATLAB and lp_solve IDE program. The performance evaluation results show that end-to-end throughput is improved by using multirate and intra-session network coding can achieve better throughput than opportunistic routing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.895-906
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• 1993

In this paper, MTSVQ(Multirate Tree-Structured Vector Quantization) is introduced for subband image coding. Original images are decomposed into a number of subband components, and multiresolution codebook is designed by MTSVQ algorithm. Optimal bit allocation among the subband components becomes the problem selecting the particular pruned subtree of MTSVQ which has the desired rate and distortion.

• Journal of Communications and Networks
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• v.5 no.2
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• pp.116-123
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• 2003

In this paper, we address the problem of designing multirate codes for a multiple-input and multiple-output (MIMO) system by restricting the receiver to be a successive decoding and interference cancellation type, when each of the antennas is encoded independently. Furthermore, it is assumed that the receiver knows the instantaneous fading channel states but the transmitter does not have access to them. It is well known that, in theory, minimummean- square error (MMSE) based successive decoding of multiple access (in multi-user communications) and MIMO channels achieves the total channel capacity. However, for this scheme to perform optimally, the optimal rates of each antenna (per-antenna rates) must be known at the transmitter. We show that the optimal per-antenna rates at the transmitter can be estimated using only the statistical characteristics of the MIMO channel in time-varying Rayleigh MIMO channel environments. Based on the results, multirate codes are designed using punctured turbo codes for a horizontal codedMIMOsystem. Simulation results show performances within about one to two dBs of MIMO channel capacity.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.49-52
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• 2000

The Multi-Code CDMA systems that are proposed as an effective transmission methodology in the IMT-2000 systems allow higher rate services under the IS-95 CDMA infrastructure. The Multi-Code CDMA systems convert the higher rate data into the lower rate by serial to parallel operation and spread the converted data streams by the multiple walsh codes, and its mobile receiver needs multiple walsh generators and data correlators to demodulate simultaneously multiple walsh code channels. Therefore, the number of data correlators is increased as the number of traffic channels increases. In this paper, we proposed the new structure of the data correlators using walsh overlay coding, the shared accumulator, and FWHT(Fast Walsh Hadamard Transform) algorithm for reducing the bottle-neck effect resulting the increase of the number of data correlators.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2A
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• pp.168-175
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• 2006

To cope with channel variation, wireless networks such as IEEE 802.11 WLAN provide multiple transmission rates by employing different channel modulation and coding schemes. However, the coexistence of different transmission rates degrades the total system performance of the network. In order to eliminate this performance abnormality and improve protocol capacity, we propose a new Packet transmission algorithm, the RAT(Rate-Adapted Transmission) scheme. The RAT scheme distributes the wireless channel fairly based on the channel occupancy time. Moreover, it efficiently transmits packets even in a single station using rate-based queue management. Therefore, the RAT scheme obtains not only the inter-rate contention gain among stations but also the intra-rate contention gain among connections in a single station.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10C
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• pp.1451-1459
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• 2004

In this paper, a low-power 32-point IMDCT structure is proposed for MP3. Through re-odering of IMDCT matrices, we propose the systolic structure operating with 16, 8, 4, 2, and 1 cycle, respectively. To reduce power consumption, multiplication of each sub blocks are implemented by add and shift operation with CSD(Canrmic sigled digit) form coefficients. To reduce, furthermore, the number of adders, we utilize the common sub-expression sharing techniques. With these techniques, the relative power consumption of the proposed structure is reduced by 58.4% comparison to the conventional structure using only 2's complement form coefficient. Validity of the proposed structure is proved through Verilog-HDL coding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.5
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• pp.963-969
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• 2000

In this paper, we define two input-output gains of linear periodic time-varying systems. One is the ratio of output with worst-case l2-norm over all inputs with unit 12-norm. It denotes G($\iota_2,\iota_2$.The other is the ratio of output with worst-case RMS value over all inputs with unit RMS value. It denotes G(RMS, RMS) .It is fact that these two gains are equivalent for linear time-invariant system. In this paper, we prove these two gains are also equivalent for linear periodic time-varying system. In addition, the relationship between two method of obtaining the generalized frequency responses for linear periodic time-varying system is derived. Finally, we apply the defined input-output gains to M-channel filter-bank which is multi-rate signal Processing system, used to speech coding. In the filter-bank, generally, aliasing distortion, magnitude distortion, and phase distortion are present. It is shown that these are kept small if the filter-bank is designed by a method that optimizes the gain G($\iota_2,\iota_2$ of an error system.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.52-57
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• 2005

The adaptive multirate wideband (AMR-WB) speech coder has an extended audio bandwidth from 50 Hz to 7 kBz and operates on nine speech coding bit-rates from 6.6 to 23.85 kbit/s. In this Paper, we present the real-time implementation of AMR-WB speech coder using 16bit fixed-point TMS320C5509 that has dual MAC units. Firstly, We implemented AMR-WB speech coder in C 1anguage level using intrinsics, and then performed optimization in assembly language. The computational complexity of the implemented AMR-WB coder at 23.85 kbit/s is 42.9 Mclocks. And this coder needs the program memory of 15.1 kwords, data ROM of 9.2 kwords and data RAM of 13.9 kwords.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.