• Journal of the Korea Institute of Information Security & Cryptology
• /
• v.23 no.4
• /
• pp.609-615
• /
• 2013

This paper represents the development of the HD-Voice application with G.711.1 coder-the latest wideband codec standard from ITU-T-for smartphone based on android platform. The work also includes the structure of the HD-voice application and the result of speech quality of HD-Voice application with G.711.1 coder. The paper shows that the speech quality of HD-Voice application with G.711.1 coder is excellent.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.47 no.5
• /
• pp.62-73
• /
• 2010

In this paper we proposed a candidate algorithm on G.711.1 and G.722 superwideband extension codec which is under standardization by ITU-T. The proposed codec not only provides an interoperable bitstream with ITU-T G.711.1 and G.722, but also encodes a superwideband signal with a bandwidth of 50-14,000 Hz using superwideband extension layer. The candidate codec consists of a core layer to provide an interoperability with conventional wideband codecs and superwideband extension layer using linear prediction-based sinusoidal coding. The proposed extension codec operates on 5ms frame and provides four superwideband bitrates of 64, 80, 96, and 112 kbit/s depending on the core codec. Since the resulting bitstream has an embedded structure, it can be converted into core bitstream by simple truncation without transcoding. The proposed codec has a short algorithmic delay and low complexity and passed the qualification test of G.711.1 and G.722 superwideband extension codec performed by ITU-T.

• MALSORI
• /
• no.57
• /
• pp.175-189
• /
• 2006

Because a packet loss brings about degradation of speech quality, VoIP speech coders have PLC (Packet Loss Concealment) mechanism. G.711, which is a mandatory VoIP speech coder, also has the PLC algorithm based on pitch period replication. However, it is not robust to burst losses. Thus, we propose two methods to improve the performance of the original PLC algorithm in G.711. One adaptively utilizes voiced/unvoiced information of adjacent good frames regarding to the current lost frame. The other is based on adaptive gain control according to energy variation across the frames. We evaluate the performance of the proposed PLC algorithm by measuring a PESQ value under different random and burst packet loss simulating conditions. It is shown from the experiments that the performance of the proposed PLC algorithm outperforms that of PLC employed in ITU-T Recommendation G.711.

• The Journal of the Acoustical Society of Korea
• /
• v.34 no.5
• /
• pp.403-409
• /
• 2015

In this paper, we propose Packet Loss Concealment (PLC) method using adaptive signal scale estimation for performance improvement of G.711 PLC. The conventional method controls a gain using 20 % attenuation factor when continuous loss occurs. However, this method lead to deterioration because that don't consider the change of signal. So, we propose gain control by adaptive signal scale estimation through before and after frame information using Least Mean Square (LMS) predictor. Performance evaluation of proposed algorithm is presented through Perceptual Evaluation of Speech Quality (PESQ) evaulation.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2008.02a
• /
• pp.125-128
• /
• 2008

IPTV의 상용화에 맞추어 사용자와 TV간의 정보 교환에 의한 대화형 서비스들이 제공되고 있으며, 특히 음성인식 기술은 이러한 서비스를 실현하기 위한 중요한 기술 중의 하나로 대두되고 있다. TV에서의 음성인식 수행을 위해서는 가정환경과 같은 제한된 공간에서 효율적으로 사용자의 음성을 TV에 전송할 수 있는 근거리 무선통신 수단이 필요하게 된다. 특히, 리모트 컨트롤러와 같은 저전력 시스템 환경에서 구현이 가능해야 한다. 따라서 이러한 제한된 조건에서 최적의 성능을 갖는 음성 전송 시스템 개발이 요구되고 있다. 본 논문에서는 블루투스 환경 하에서 음성인식을 위해 필요한 음성전송 시스템을 실시간 구현한다. 효율적인 음성전송을 위해 G.711을 기본 코덱으로 사용하며, 음성전송 시 발생하는 패킷손실에 따른 음성 품질 저하를 줄이기 위해 G.711 패킷손실 은닉 알고리즘을 음성전송 시스템에 적용한다. 특히 G.711 패킷 손실 은닉 알고리즘 수행을 위해 블루투스 프로토콜 스택application layer에 RTP 프로토콜을 적용하여 패킷 손실 여부를 확인하고, 패킷 손실 발생 시 패킷손실 은닉 알고리즘을 통해 음성의 품질 저하를 줄인다. 구현된 시스템의 성능을 평가한 결과, G.711 패킷 손실 알고리즘을 적용하여 2~10%의 패킷손실 환경에서 14.7%의 음질개선을 얻을 수 있었다.

• 한국HCI학회:학술대회논문집
• /
• 2008.02a
• /
• pp.218-223
• /
• 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%.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.37 no.7B
• /
• pp.528-535
• /
• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• ETRI Journal
• /
• v.35 no.3
• /
• pp.542-545
• /
• 2013

State-of-the-art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model-based coding algorithm of MDCT spectral coefficients, apply it to the ITU-T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.

• 한국신재생에너지학회:학술대회논문집
• /
• 2005.11a
• /
• pp.711-711
• /
• 2005

• Korean Journal of Agricultural Science
• /
• v.25 no.1
• /
• pp.124-130
• /
• 1998

An alkalophilic bacterium, the strain AC-711 as a potent producer of alkaline cellulase, was selected among many isolates from soil environments. Morphological, physiological and chemical characteristics of the strain AC-711 suggested that it belongs to the genus Pseudomonas according to the Bergey's Manual of Systematic Bacteriology, however the G+C mol% (54.43) of its chromosomal DNA is lower than the normal values of the genus. The major cell wall fatty acids were determined as 15:0 and 17:0 anteiso. The production of alkaline CMCase by the strain was maximal when grown on the mediun containing 1% carboxymethyl cellulose, 0.1% $KH_2PO_4$, 0.02% $CoCl_2$, 0.02% Tween 80, 0.5% $Na_2CO_3$, 0.8% yeast extract, pH 10.3 at $30^{\circ}C$ for 3 days, and the most of enzyme was excreted into culture mediun.