• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.102-108
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• 2007

A high-Performance scalable speech codec generally requires a very low-rate first layer and a fine granule second layer, and this codec can be implemented with the harmonic codec and the MDCT-based transform codec for each layer. In this structure, however. each codec requires independent frequency transform and the time delay of each codec is accumulated. resulting in long time delay for the overall codec. In this paper, new MDCT structure in the second layer is Proposed. where MDCT is forced to share the look-ahead region of the first layer in order to prevent the time delay accumulation and the resulting functional error of MDCT is analyzed and removed after IMDCT The Proposed delayless MDCT requires no additional bits and Provides the equivalent coding performance with the reduced time delay, yielding a meaningful enhancement of the overall codec.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.2
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• pp.369-374
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• 2018

The Line Code is a code used for data transmission and is a coding method used to prevent loss of data between transmitter and receiver. There are various kinds of line code such as B3ZS, HDB3, B8ZS, and HDB3 code is used in the tactical communication networks of Korea. In this paper, we implement the HDB3 Codec that is used in tactical communication networks in order to eliminate the risk factors for component discontinuance and to shorten development period and ensure reliability. Also it is confirmed that it can be implement by minimizing the amount of logic usage so that it can be used unrestrictedly in systems using FPGA and the implemented HDB3 Codec is simulated to confirm that it is equivalent to the performance of HDB3 Codec IC.

• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.4
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• pp.117-125
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• 2014

In this paper, a novel hardware architecture of the LT codec is presented where non-BP based decoding algorithm is applied. Novel LT codec architecture is designed with an efficient degree distribution unit using Verilog HDL. To perform permutation operation, different initial valued or time shifted counters have been used to get pretty well permutations and an effect of randomness. The codec will take 128 bits as input and produce 256 encoded output bits. The simulation results show expected performances as the implemented distribution and the original distribution are pretty same. The proposed LT codec takes 257.5 cycle counts and $2.575{\mu}s$ for encoding and decoding instead of 5,204,861 minimum cycle counts and 4.43s of the design mentioned in the previous works where iterative soft BP decoding was used in ASIC and ASIP implementation of the LT codec.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.143-146
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• 1998

A new stereoscopic codec. structure using MPEG-2 multiview profile is presented in this paper. In the suggested codec., the left image is coded with motion estimation in the base layerand the right image is coded with disparity estimation in the enhancement layer. Since it is possible to calculate rough motion of the right image sequence with disparity and motion of the left image sequence, motion compensation of the enhancement layer is performed without motion estimation. Since the proposed codec. does not perform motion estimation in the enhancement layer encoding, it is simple and reduces the encoding time. We compared the PSNR of encoded image with three different structured codec., and the experimental results show that suggested codec. has comparable with other codecs.

• 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.

• IEIE Transactions on Smart Processing and Computing
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• v.4 no.6
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• pp.434-442
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• 2015

In this paper, we propose an integrated spatial and view scalable video codec based on high efficiency video coding (HEVC). The proposed video codec is developed based on similarity and uniqueness between the scalable extension and 3D multi-view extension of HEVC. To improve compression efficiency using the proposed scalable multi-view video codec, inter-layer and inter-view predictions are jointly employed by using high-level syntaxes that are defined to identify view and layer information. For the inter-view and inter-layer predictions, a decoded picture buffer (DPB) management algorithm is also proposed. The inter-view and inter-layer motion predictions are integrated into a consolidated prediction by harmonizing with the temporal motion prediction of HEVC. We found that the proposed scalable multi-view codec achieves bitrate reduction of 36.1%, 31.6% and 15.8% on the top of ${\times}2$, ${\times}1.5$ parallel scalable codec and parallel multi-view codec, respectively.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8C
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• pp.575-582
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• 2008

In this paper we described the process and the results of real-time implementation of G.729.1 wideband speech codec which is standardized in SG15 of ITU-T. To apply the codec on ARM926EJ-S(R) processor core. we transformed some parts of the codec C program including basic operations and arithmetic functions into assembly language to operate the codec in real-time. G.729.1 is the standard wideband speech codec of ITU-T having variable bit rates of $8{\sim}32kbps$ and inputs quantized 16 bits PCM signal per sample at the rate of 8kHz or 16kHz sampling. This codec is interoperable with the G.729 and G.729A and the bandwidth extended wideband($50{\sim}7,000Hz$) version of existing narrowband($300{\sim}3,400Hz$) codec to enhance voice quality. The implemented G.729.1 wideband speech codec has the complexity of 31.2 MCPS for encoder and 22.8 MCPS for decoder and the execution time of the codec takes 11.5ms total on the target with 6.75ms and 4.76ms respectively. Also this codec was tested bit by bit exactly against all set of test vectors provided by ITU-T and passed all the test vectors. Besides the codec operated well on the Internet phone in real-time.

• 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.