• The Journal of the Acoustical Society of Korea
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• v.37 no.2
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• pp.110-117
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• 2018

In this paper, we propose a low delay window switching MDCT (Modified Discrete Cosine Transform) method for speech/audio coder. The window switching algorithm is used to reduce the degradation of sound quality in non-stationary trasient duration and to reduce the algorithm delay by using the low delay TDAC (Time Domain Aliasing Cancellation). While the conventional window switching algorithms uses overlap-add with different lengths, the proposed method uses the fixed overlap add length. It results the reduction of algorithm delay by half and 1 bit reduction in frame indication information by using 2 window types. We apply the proposed algorithm to G.729.1 based on MDCT in order to evaluate the performance. The propose method shows the reduction of algorithm delay by half while speech quality of the proposed method maintains same as the conventional method.

• Journal of Power Electronics
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• v.16 no.1
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• pp.297-309
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• 2016

Grid-connected inverters (GCIs) with an LCL output filter have the ability of attenuating high-frequency (HF) switching ripples. However, by using only grid-current control, the system is prone to resonances if it is not properly damped, and the current distortion is amplified significantly under highly distorted grid conditions. This paper proposes a synchronous reference frame equivalent proportional-integral (SRF-EPI) controller in the αβ stationary frame using the parallel virtual resistance-based active damping (PVR-AD) strategy for grid-interfaced distributed generation (DG) systems to suppress LCL resonance. Although both a proportional-resonant (PR) controller in the αβ stationary frame and a PI controller in the dq synchronous frame achieve zero steady-state error, the amplitude- and phase-frequency characteristics differ greatly from each other except for the reference tracking at the fundamental frequency. Therefore, an accurate SRF-EPI controller in the αβ stationary frame is established to achieve precise tracking accuracy. Moreover, the robustness, the harmonic rejection capability, and the influence of the control delay are investigated by the Nyquist stability criterion when the PVR-based AD method is adopted. Furthermore, grid voltage feed-forward and multiple PR controllers are integrated into the current loop to mitigate the current distortion introduced by the grid background distortion. In addition, the parameters design guidelines are presented to show the effectiveness of the proposed strategy. Finally, simulation and experimental results are provided to validate the feasibility of the proposed control approach.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.48-55
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• 2010

It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.

• Journal of Sensor Science and Technology
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• v.16 no.2
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• pp.126-131
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• 2007

In this paper, we fabricated piezoresistive pressure sensor with dry etching technology which used ICP-RIE (inductively coupled plasma reactive ion etching) and etching delay technology which used SOI (silicon-on-insulator). Structure of the fabricated pressure sensor shows a square diaphragm connected to a frame which was vertically fabricated by dry etching process and a single-element four-terminal gauge arranged at diaphragm edge. Sensitivity of the fabricated sensor was about 3.5 mV/V kPa at 1 kPa full-scale. Measurable resolution of the sensor was not exceeding 20 Pa. The nonlinearity of the fabricated pressure sensor was less than 0.5 %F.S.O. at 1 kPa full-scale.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2005.06a
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• pp.1013-1017
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• 2005

This paper proposes an effective algorithm, P-DCF(Polling DCF) to satisfy real-time requirements in the industrial environments, based on 802.11 DCF protocols. Theoretical analysis and actual experiments are carried out, which shows that the proposed P-DCF brings several benefits: 1) it reduces the time delay for frame transmission 2)it guarantees bounded upper limit of maximum transmission delay 3) it maintains fairness

• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2806-2808
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• 2005

The Controller Area Network (CAN) is being widely used for real-time control application and small-scale distributed computer controller systems. When the stuff bits are generated by bit-stuffing mechanism in the CAN network, it causes jitter including variations in response time and delay. In order to eliminate this jitter, stuff bit must be controlled to minimize the response time and reduce the variation of data transmission time. At first, this paper shows that conventional CAN protocol causes the transmission time delay. Secondly, this paper proposes the method to reduce the stuff bits by restriction of available identifier. Finally, data manipulation method can be reduced the number of stuff-bits in the data field. The proposed restriction method of ID and manipulating data field are pretty useful to the real-time control strategy with respect to performance. These procedures are implemented in local controllers of the ISHURO (Inha Semyung Humanoid Robot).

• Journal of Korean Society of Industrial and Systems Engineering
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• v.28 no.1
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• pp.129-133
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• 2005

Slotted ALOHA(S-ALOHA) is widely used in local wireless network. We analyze the performance of contention-based model in wireless LAN using S-ALOHA protocol. We analyze the performance of binary exponential backoff (BEB) algorithm under the slotted ALOHA protocol: whenever a node's message which tries to reserve a channel is involved in a collision for the ith time, it chooses one of the next $2^i$ frames with equal probability and attempts the reservation again. We derive the expected access delay and throughput which is defined as the expected number of messages that reserve a channel in a frame. A simulation study is performed to verify our method.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.257-258
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• 2007

In real-time video communications, the large fluctuation of bits generated from contiguous frames may lead to the buffer oveflow and the delay of transmitting video data. This paper proposes a method that the bits generated from each frame may be in the range of specified fluctuation. The method uses a scene change detector and ignores the time information of detected frames. The experimental results indicate that our scheme achieves average PSNR gain of 2.0dB over TMN8.

• MALSORI
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• no.52
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• pp.111-120
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.