• Proceedings of the IEEK Conference
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• 2001.06b
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• pp.333-336
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• 2001

This paper describes the design and implementation results of 40, 32, 24 and 16kbps ADPCM encoder and decoder circuit, based on the protocol CCITT G.726. We verified the ADPCM algorithm using C language and designed the RTL circuit with Verilog HDL. The circuit has been simulated by Verilog-XL, synthesized by Design Compiler and verified using Xilinx FPGA. Since the synthesized circuit includes a small number of gates, it is expected to be used as a core module in the digital voice and image processing SOC.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.151-154
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• 1998

이중 전송율(5.3 과 6.3kbit/s)을 제공하는 G.723.1 음성 코더는 공중망을 통한 H.324 POTS 영상 회의 규격의 음성 코더로 채택된 것으로, MPMLQ, ACELP 알고리즘에 근거한다. 본 논문에서는 Annex A를 포함한 G.723.1 음성 코더 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.723.1 에 대한 양방향 평가가 Codec loopback을 통해 수행되었으며, ITU에서 제공한 테스트 절차에 따라 평가되었다. 또한, 본 논문에서 구현된 G.723.1 음성 코더는 27MIPS의 계산 속도를 갖으며, 프로그램 ROM의 크기는 8.85K Words이고, 10K 데이터 ROM과 4K 데이터 RAM을 필요로 하고 있다. 경쟁 제품과의 MOS 측정 음질 평가를 실시한 결과, CSD17C00에서의 음질 성능이 더 우수함을 입증 함으로써, 본 논문에서 보여준 CSD17C00을 기반으로 구현된 G.723.1 알고리즘의 실시간 구현기술의 타당성을 검증하게 되었다.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.811-814
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• 2000

It this paper a VLSI architecture of the Shape-Adaptive Discrete Cosine Transform (SA-DCT) is described, which can be employed dedicatedly for MPEG-4 video codec. Adopting a fast DCT algorithm, the number of multipliers can be reduced by half in comparison with a conventional algorithm. This SA-DCT core with a small additional amount of hardware can perform the SA-Inverse DCT (SA-IDCT) by sharing multipliers and a transportation memory. The proposed SA-DCT core is integrated with 40,000 gates by using 0.35$mu$m triple-metal CMOS technology, which operates at 20 Mhz, and hence enables the realtime codec of CIF ($352{\times}288$ pixels) pictures.

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

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.233-241
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• 2020

In this paper, a new buffer control method is proposed for improving the performance of the frequency domain part of the 3D audio (3DA) core codec. For the proposed buffer control method, we first combine the 3DA RM9 with the 3GPP AAC buffer control method which includes the psychoacoustic model and rate-distortion control process with the spectral hole avoidance algorithm. Then, we revise the 3GPP buffer control method so as to achieve a faithful bit allocation to the frames with higher activity. With the MUSHRA test, we prove that the proposed buffer control method demonstrates better performance than the 3DA RM9 and 3GPP AAC.

• Information and Communications Magazine
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• v.25 no.12
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• pp.10-18
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• 2008

In this paper, we present a low-power design of H.264 codec based on dedicated hardware and software solution on EMP(ETRI Multi-core platform). The dedicated hardware scheme has reducing computation using motion estimation skip and reducing memory access for motion estimation. The design reduces data transfer load to 66% compared to conventional method. The gate count of H.264 encoder and the performance is about 455k and 43Mhz@30fps with D1(720x480) for H.264 encoder. The software solution is with ASIP(Application Specific Instruction Processor) that it is SIMD(Single Instruction Multiple Data), Dual Issue VLIW(Very Long Instruction Word) core, specified register file for SIMD, internal memory and data memory access for memory controller, 6 step pipeline, and 32 bits bus width. Performance and gate count is 400MHz@30fps with CIF(Common Intermediated format) and about 100k per core for H.264 decoder.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.77-78
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• 2008

With advent of 3G mobile communication services, video telephony becomes one of the major services. However, due to a narrow channel bandwidth, the current video telephony services have not yet reached a satisfied level. In this paper, we propose an ROI (Region-Of-Interest) based improvement of visual quality for video telephony services with the H.264|MPEG-4 Part 10 (AVC: Advanced Video Coding) codec. To this end, we propose a face detection and tracking method to define ROI for the AVC codec based video telephony. Experiment results show that our proposed ROI based method allowed for improved visual quality in both objective and subjective perspectives.

• ETRI Journal
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• v.25 no.6
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• pp.489-502
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• 2003

This paper presents an MPEG-4 video codec, called MoVa, for video coding applications that adopts 3G-324M. We designed MoVa to be optimal by embedding a cost-effective ARM7TDMI core and partitioning it into hardwired blocks and firmware blocks to provide a reasonable tradeoff between computational requirements, power consumption, and programmability. Typical hardwired blocks are motion estimation and motion compensation, discrete cosine transform and quantization, and variable length coding and decoding, while intra refresh, rate control, error resilience, error concealment, etc. are implemented by software. MoVa has a pipeline structure and its operation is performed in four stages at encoding and in three stages at decoding. It meets the requirements of MPEG-4 SP@L2 and can perform either 30 frames/s (fps) of QCIF or SQCIF, or 7.5 fps (in codec mode) to 15 fps (in encode/decode mode) of CIF at a maximum clock rate of 27 MHz for 128 kbps or 144 kbps. MoVa can be applied to many video systems requiring a high bit rate and various video formats, such as videophone, videoconferencing, surveillance, news, and entertainment.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.618-629
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• 2009

The MPEG-4 Audio Lossless Coding (ALS) standard belongs to the family MPEG-4 audio coding standards. In contrast to lossy codecs such as AAC, which merely strive to preserve the subjective audio quality, lossless coding preserves every single bit of the original audio data. The ALS core codec is based on forward-adaptive linear prediction, which combines remarkable compression with low complexity. Additional features include long-term prediction, multichannel coding, and compression of floating-point audio material. This paper describes the basic elements of the ALS codec with a focus on prediction, entropy coding, and related tools and points out the most important applications of this standardized lossless audio format.