• Journal of Broadcast Engineering
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• v.21 no.2
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• pp.219-236
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• 2016

In this paper, we design and implement a 3D scalable video codec by combining the Scalable HEVC (SHVC) and the 3D-HEVC which are the extended standards of High Efficiency Video Coding (HEVC). The proposed 3D scalable video codec supports the view and spatial scalabilities which are the properties of 3D-HEVC and SHVC, respectively. In the proposed 3D scalable codec, the high-level syntaxes are designed to support the multiple scalabilities. In the computer simulation section, we confirmed the conformance of the proposed codec and analyzed the performance of the proposed codec.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.397-399
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• 2017

The HDB3 code, a type of line code, is a data coding method used for digital data transmission. It is used to remove the DC wander on the transmission line which occurs when the DC component data is transmitted continuously. The military tactical communication network uses HDB3 code for data transmission and develops equipment using commercial HDB3 Codec IC. Because it is operated for more than 10 years due to the characteristics of military equipment, if a failure occurs in the equipment, the equipment can not be repaired due to the discontinuance of the part, so that the entire equipment may not be used. In this paper, we implement the HDB3 Codec as a logic to solve this problem and verify that the performance is equivalent to that of commercial parts.

• 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 the Acoustical Society of Korea
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• v.22 no.4
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• pp.308-314
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• 2003

Wideband speech communications with 7㎑ bandwidth can provide high-quality speech services that are almost impossible with current narrow-band speech communications with 3.4 ㎑ bandwidth, and AMR wideband codec was recently developed for these services. The performance of AMR wideband codec is excellent due to its wideband information and partially to ACELP structure, but it requires high computational complexity especially in codebook search. In this paper, to solve this problem, an efficient codebook search method for AMR wideband codec is proposed. The proposed method first determines the coarse initial codevector, then improves the performance of codevector by replacing a poor pulse in codevector with better one iteratively. Simulations show that AMR wideband codec with proposed codebook search method has higher performance with much less computational cost than conventional AMR wideband codec.

• Journal of Korea Society of Industrial Information Systems
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• v.23 no.6
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• pp.59-66
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• 2018

In this paper, we experimented how long distance voice communication is possible After implemented PTT(Push to talk) Bi-directional radio using open source project Codec 2, which is a low speed voice codec for digital amateur radio and 900MHz FSK transceiver. In case of a general digital radio, the AMBE+2 codec, which is regarded as an industry standard in terms of performance, is expensive and has the monopoly of technology. Using the 400MHz band in terms of frequency, narrow bandwidth of DMR(12.5kHz) and DPMR(6.25kHz) is used, so the data rate is low. In the 900MHz bandwidth can be extended, which is advantageous in terms of data transmission. As a result of the voice quality and distance field test, we could find that the communication takes place within about 500m. In this paper, only voice communication is reviewed. if a review of data transmission such as a simple image is added, this solution can be used in various fields as a low cost IOT radio.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.2
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• pp.366-385
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• 2013

In this paper, we propose a compressed sensing (CS) based Wyner-Ziv (WZ) codec using motion-aligned auto regressive model (MAAR) based side information (SI) extrapolation to improve the compression performance of low-delay distributed video coding (DVC). In the CS based WZ codec, the WZ frame is divided into small blocks and CS measurements of each block are acquired at the encoder, and a specific CS reconstruction algorithm is proposed to correct errors in the SI using CS measurements at the decoder. In order to generate high quality SI, a MAAR model is introduced to improve the inaccurate motion field in auto regressive (AR) model, and the Tikhonov regularization on MAAR coefficients and overlapped block based interpolation are performed to reduce block effects and errors from over-fitting. Simulation experiments show that our proposed CS based WZ codec associated with MAAR based SI generation achieves better results compared to other SI extrapolation methods.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.76-82
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• 2001

In this paper, a novel quantization method for wideband signal codec with 7 kHz bandwidth is proposed. In the transform-based wideband signal codecs, the signal is transformed to frequency domain and the spectral coefficients in each frequency band are quantized based on human perceptual model, followed by Huffman coding. However, the property of each band varies with frequency, and the codec has poor performance when all bands are quantized with the same method. Therefore, a selective quantization method is proposed, which analyzes the band property and selects the quantization domain between frequency domain and time domain based on the quantization efficiency. It is confirmed that the proposed method has better performance than the quantizer of G722.1 codec.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.1
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• pp.71-76
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• 1997

In this paper, we describe simulation results of subjective assessments with bit rate variation of multi-channel audio codec system for the services of HDTV satellite broadcasting services. Based on this experiment results, we also describe the specification and subjective performance results for 4-channel audio codec. For multi-channel, bit rates are 384,320,256,128kbps and the results show that 320kbps bit rate is needed to compare with the original and the reproduced signal. Based on this, for 384kbps for 4-channel audio codec, three items that achieve a diffgrade worse than -0.5 are due to the noise of analog output module. This system is satisfied for the audio codec of the HDTV system.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.2
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• pp.93-100
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• 2009

As the Internet Protocol be widely/rapidly used in packet communication, common carriers are providing the multimedia service(Both direction real-time voice, video conference, remote educational etc.)on the Internet. Also the 070 VoIP (Voice over IP) service is provided by the carriers on the packer network. In order to offer VoIP service in Korea, common carrier has to acquire the optimum level for QoS(quality of service). In this paper, we study on CODEC quality to get a higher QoS for VoIP.