• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.89-92
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• 2001

In this paper, adaptive multi-stage parallel interference cancellation (PIC) receiver is considered for a multi-rate DS-CDMA system. In each stage of the adaptive multi-stage PIC receiver, multiple access interference (MAI) estimates are obtained using the sub-bit estimates from the Previous stage and the adaptive weights for the sub-bit estimates. The adaptive weights are obtained by minimizing the mean squared error between the received signal and its estimate through a least mean square (LMS) algorithm. It is shown that the adaptive multi- stage PIC receiver achieves smaller BER than the matched filter receiver, multi-stage PIC receiver, and multi-stage partial PIC receiver for the multi-rate DS-CDMA system in a Rayleigh fading channel.

• Journal of Communications and Networks
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• v.7 no.4
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• pp.462-470
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• 2005

We propose a multi-rate and multi-BEP transmission scheme using adaptive overlapping pulse-position modulator (OPPM) and optical power controller in optical code division multiple access (CDMA) networks. The proposed system achieves the multi-rate and multi-BEP transmission by accommodating users with different values of OPPM parameter and transmitted power in the same network. The proposed scheme has advantages that the system is not required to change the code length and number of weight depending on the required bit rate of a user and the difference of bit rates does not have so much effect on the bit error probabilities (BEPs). Moreover, the difference of transmitted powers does not cause the change of bit rate. We analyze the BEPs of the four multimedia service classes corresponding to the com­binations of high/low-rates and low/high-BEPs and show that the proposed scheme can easily achieve distinct differentiation of the service classes with the simple system configuration.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.396-402
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• 2003

본 논문에서는 multi-media service를 제공하는데 smart antenna system을 적용했을 때의 performance 를 각 서비스별 특성에 따라 평가한다. 서로 다른 다중 rate 의 신호가 각각의 안테나를 통해서 수신되는 환경 하에서 adaptive smart antenna를 적용함으로써 보다 고속이고 향상된 용량의 서비스를 제공할 수 있도록 하기 위해서는 사용자의 지리적 분포 또는 간섭 전력에 따라서 성능 특성이 달라지므로 이에 대한 성능 분석을 시뮬레이션을 통해서 수행한다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.755-758
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• 2000

본 논문에서는 AMR(Adaptive Multi-Rate)를 이용하여 광대역 음성부호화기를 설계하였다. 16kHz로 샘플링 된 입력 신호를 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 decimation하여 두 개의 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz -7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)과 ATC(Adaptive Transform Coding)을 사용하여 각각 부호화되어 전송된다. 두 대역으로부터 부호화된 정보는 20.2kbps에서 12.75kbps까지의 전송률을 갖고, 수신단에서는 각 대역을 AMR과ATC방법으로 역부호화하여 음성신호를 합성한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위해 ITU-T의 표준안인 G.722를 포함하여 MOS 시험을 하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.5B
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• pp.632-638
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• 2001

본 논문에서는 AMR(Adaptive Multi-Rate)를 이용하여 광대역 음성부호화기를 설계하였다. 16kHz로 샘플링된 입력 신호를 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 decimation하여 두 개의 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz∼7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)과 ATC(Adaptive Transform Coding)을 사용하여 각각 부호화되어 전송된다. 두 대역으로부터 부호화된 정보는 20.2kbps에서 12.75kbps까지의 전송률을 갖고, 수신단에서는 각 대역을 AMR과 ATC 방법으로 역부호화하여 음성신호를 합성한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위해 ITU-T의 표준안인 G.722를 포함하여 MOS 시험을 하였다.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.827-830
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• 2000

3GPP and ETSI adopted AMR(Adaptive Multi-Rate) as a standard for next generation IMT-2000 service. In this paper, we analyzed algorithm about AMR and optimized ANSI C source on the C complier and assembly language of Texas Instrument . The implemented AMR speech codec requires 28.2MIPS of complexity for encoder and 5.5MIPS for decoder. we performed real-time implementation of AMR speech codec using 82% of TMS320C5402 with 40 MIPS specification. We give proof that the output speech of the implemented speech codec on DSP board is identical with result of C source program simulation. Also the reconstructed speech is verified in the real-time environment consisted of microphone and speaker.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.56-61
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• 2001

This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.

• International Journal of Naval Architecture and Ocean Engineering
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• v.9 no.6
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• pp.668-676
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• 2017

A two-dimensional numerical investigation is performed to study the influence of slot width of multi-stage stationary floating overtopping wave energy devices on overtopping flow rate and performance. The hydraulic efficiency based on captured crest energy of different device layouts is compared with that of single-stage device to determine the effect of the geometrical design. The results show optimal trends giving a huge increase in overtopping energy. Plots of efficiency versus the relative slot width show that, for multi-stage devices, the greatest hydraulic efficiency is achieved at an intermediate value of the variable within the parametric range considered, relative slot width of 0.15 and 0.2 depending on design layouts. Moreover, an application of adaptive slot width of multi-stage device is investigated. The numerical results show that the overall hydraulic efficiency of non-adaptive and adaptive slot devices are approximately on par. The effect of adaptive slot width on performance can be negligible.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.11
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• pp.4519-4533
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• 2015

Hypertext Transfer Protocol (HTTP) adaptive streaming has become a new trend in video delivery. An HTTP adaptive streaming client needs to effectively estimate resource availability and demand. However, due to the bitrate of the video encoded in variable bitrate (VBR) mode, a bitrate mismatch problem occurs. With the rising demand for mobile devices, the likelihood of cases where two or more HTTP adaptive streaming clients share the same network bottleneck and competing for available bandwidth will increase. These mismatch and competition issues lead to network congestion, which adversely affects the Quality of Experience (QoE). To solve these problem, we propose a video rate adaptation scheme for the HTTP video streaming to guarantee and optimize the QoE. The proposed scheme estimates the available bandwidth according to the bitrate of each segment and also schedules the segment request time to expedite the response to the bandwidth variation. We used a multi-client simulation to prove that our scheme can effectively cope with drastic changes in the connection throughput and video bitrate.