• Electronics and Telecommunications Trends
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• v.10 no.4 s.38
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• pp.129-142
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• 1995

본 고에서는, 음성을 디지털로 부호화하여 전송함으로써 발생되는 신호 대 양자화왜곡 비(Q)의 개념 및 CODEC과의 관계를 분석하고, MNRU를 디지털 회로로 구현하는데 필요한 입력음성 신호레벨, 잡음의 통계적 성질 및 진폭제한이 음성품질에 미치는 영향을 살펴보았다. 또한, 본 연구에서 구현한 MNRU의 성능에 대해 주관평가 실험을 실시하여, 다른 나라의 주관평가 결과와 비교/분석하였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.96-103
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• 2000

The voice traffic is one of the most important real-time objects in WATM(Wireless Asynchronous Transfer Mode) networks. In this paper, we propose a new MAC(Medium Access'Control) protocol for the efficienttransmission of voice traffic over WATM networks in the multimedia environment and compare the performanceto existing similar protocols. The new protocol separates the reservation slot period for voice and that for data toguarantee some level of QoS(Quality of Service) in voice traffic. This is denoted by a slot assignment functiondepending on the frame size. According to the characteristics of voice traffic which is repeatedly in silent states,the protocol allocates voice reservation request slots dynamically with respect to the number of silent(off state)voice sources and also sends the first block of talkspurt restarted after silent period with a reservation requestslot to reduce the access delay.The simulation results show that the proposed protocol has better performance than Slotted Aloha in bandwidthefficiency, and can serve a certain level of QoS by the given slot assignment function even when the number ofvoice terminals varies dynamically. This means we can observe that the new MAC protocol is much better thanthe NC-PRMA(None Collision-Packet Reservation Multiple Access) protocol.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Proceedings of the KIEE Conference
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• 2002.07d
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• pp.2316-2318
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• 2002

지금까지 디지털 통신에서는 반향이 통신품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 본 논문에서는 VoIP 기반의 음성통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. 모의 실험을 통해 ECLMS 알고리즘을 적용한 반향제거기가 우수한 반향제거 성능을 보여줌을 확인하였다.

• Proceedings of the KIEE Conference
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• 2003.07d
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• pp.2127-2129
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• 2003

지금까지 디지털 통신에서는 반향이 통신 품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성 신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 현재는 DSP chip의 급속한 발달로 반향의 제거가 실시간으로 처리할수 있게 되었다. 본 논문에서는 VoIP기반의 음성 통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. DSP processor를 사용한 실험을 통해 알고리즘을 적용한 반향제거기의 성능이 우수함을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.52-57
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• 2000

In this paper, we designed the 16 Kbps speech coder that has compatibility with CS-ACELP algorithm(G.729). The speech signal is sampled at rate of 16 KHz, divided into two narrowband signal by QMF filterbank, and decimated to rate of 8 KHz. The lower-band signal is encoded by CS-ACELP and the upper-band signal is encoded by Adaptive Transform Coding(ATC) algorithm. At the receiver, two band signals are synthesized by decoder of CS-ACELP and ATC, respectively. The reconstructed output is obtained by passing the QMF synthesis bank. The proposed wideband coder is evaluated with ITU-T G.722 coder through the Mean Opinion Score(MOS) test.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.8
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• pp.35-43
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• 2004

An AAL2 transmission scheme is used to deliver voice and data traffic between Node-B and RNC on 3G WCDMA network. To predict performance of AAL2 multiplexing precisely, we derived analytically bandwidth gain and cell packing density using discrete-time Markov chain model for voice service and validated these results with simulation. We also performed detailed simulation for AAL2 multiplexing in a concentrator. Based on the analytical result, we propose the engineering guideline to select the optimal Timer_CU in a Node-B. We found that there is no major benefit of additional AAL2 multiplexing in a concentrator and the benefit of AAL2 switching in tub for data services is much less than that for voice service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.568-570
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• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.577-579
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• 2002

블루투스는 근거리 무선 인터페이스를 통만 음성 및 데이터의 전송서비스를 지원하는 통신 프로토콜이다. 블루트스와 IEEE802.11 기기들은 동일 주파수대역을 사용하므로, 간섭이 발생하고 이러만 간섭 현상은 각 기기들의 성능을 저하시킨다. 그래서, 된 논문에서 블루투스 기기를 통한 각 기기들의 Power control로 간섭문제를 완화시키는 방안을 제안한다.