• Journal of Broadcast Engineering
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• v.26 no.3
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• pp.327-330
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• 2021

In this paper, a new noise filling method is proposed for improving the performance of the 3D audio codec. In the new method, the core band is limited up to MAX_SFB, not up to the IGF start frequency. And the noise filling is applied to all frequency range of the IGF source patches. We conduct the MUSHRA test and find that the proposed noise filling method demonstrates better performance than the conventional method.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.10
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• pp.1432-1440
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• 2022

Recently, as Virtual Desktop Infrastructure(VDI) is widely used not only in office work environments but also in workloads that use high-spec multi-core-based applications, the requirements for real-time and stability of VDI are increasing. Accordingly, the display protocol used for remote access in VDI and performance optimization of virtual machines have also become more important. In this paper, we propose two ways to configure desktop virtualization for multi-core-based application operation. First, we propose a codec configuration of a display protocol with optimal performance in a high load situation due to multi-processing. Second, we propose a virtual CPU scheduling optimization method to reduce scheduling delay in case of CPU contention between virtual machines. As a result of the test, it was confirmed that the H.264 codec of Blast Extreme showed the best and stable frame, and the scheduling performance of the virtual CPU was improved through scheduling optimization.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.07a
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• pp.177-178
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• 2010

본 논문에서는 음향신호의 부호화에 있어 주부호화기로 부호화 되는 주파수 대역폭이 음질에 미치는 영향에 대하여 고찰한다. 또한 비트율을 변화시켜 양자화 잡음 발생을 줄일 수 있는 비트율의 기준에 대하여 고찰한다. 마지막으로 주관적 음질평가 및 객관적 음질평가를 통하여 그에 따른 성능을 평가한다.

• JSTS:Journal of Semiconductor Technology and Science
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• v.17 no.3
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• pp.446-457
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• 2017

In order to reduce the size of frame memory or bus bandwidth, frame memory compression (FMC) recompresses reconstructed or reference frames of video codecs. This paper proposes a novel FMC design based on discrete wavelet transform (DWT) - set partitioning in hierarchical trees (SPIHT), which supports fine-scalable throughput and is area-efficient. In the proposed design, multi-cores with small block sizes are used in parallel instead of a single core with a large block size. In addition, an appropriate pipelining schedule is proposed. Compared to the previous design, the proposed design achieves the processing speed which is closer to the target system speed, and therefore it is more efficient in hardware utilization. In addition, a scheme in which two passes of SPIHT are merged into one pass called merged refinement pass (MRP) is proposed. As the number of shifters decreases and the bit-width of remained shifters is reduced, the size of SPIHT hardware significantly decreases. The proposed FMC encoder and decoder designs achieve the throughputs of 4,448 and 4,000 Mpixels/s, respectively, and their gate counts are 76.5K and 107.8K. When the proposed design is applied to high efficiency video codec (HEVC), it achieves 1.96% lower average BDBR and 0.05 dB higher average BDPSNR than the previous FMC design.

• Journal of Broadcast Engineering
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• v.10 no.1 s.26
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• pp.57-67
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• 2005

In the paper, we propose a TV-anytime codec that encodes and decodes TV-Anytime metadata according to the TV-Anytime specification so that the resulting binary TV-Anytime metadata can be transferred efficiently through the broadcasting network where the data bandwidth is restricted.. We describe the broadcasting environment that the TV-Anytime codec will be applied to, and the required functionalities of the software modules in detail. For the design of software modules, we show how to Implement the modules for metadata fragmentation. encoding, decoding, and the fragments management. The proposed TV-Anytime codec can be utilized as the core components to a personalized digital broadcasting system providing ECG(Electronic Content Guide) and segmentation information services according to TV-Anytime standard.

• Proceedings of the IEEK Conference
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• 2004.06b
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• pp.399-402
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• 2004

This paper presents a system-on-chip prototype implementing a full integration for a portable digital audio system. The chip is composed of a audio processor block to implements audio decoding and voice compression or decompression software, a system control block including 8-bit MCU core and Memory Management Unit (MMU) a low power 16-bit ${\Sigma}{\Delta}$ CODEC, two DC-to-BC converter, and a flash memory controller. In order to support other audio algorithms except Mask ROM type's fixed codes, a novel 16-bit fixed-point DSP core with the program-download architecture is proposed. Funker, an efficient power management technique such as task-based clock management is implemented to reduce power consumption for portable application. The proposed chip has been fabricated with a 4 metal 0.25um CMOS technology and the chip area is about 7.1 mm ${\times}$ 7.1mm with 100mW power dissipation at 2.5V power supply.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.131-135
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• 2000

본 논문에서는 고성능 고정 소수점 DSP (Digital Signal Processor) 코어인 Vincent6 코어 [1]를 이용하여 ITU-T C.728 음성 부호화기를 실시간으로 구현하였다 G.728 은 16 kb/s전송률의 ITU-T표준 음성 부호화기이며, 입력신호는 8 kHz로 샘플링되며 샘플 당 16 bit 로 양자화된 PCM 신호이다. G.728 은 LD-CELP(Low Delay Code Excited Linear Prediction)라고도 하며, 알고리 듬 delay는 0.625ms 이다. Vincent6 DSP core 는 VLIW (Very-Long Instruction Word) 특성을 가지므로 다중 명령 (multiple instruction)을 수행할 수 있다 이를 위해서 G.728 annex G를 이용하여 고정 소숫점 연산으로 코드를 작성한 후, 이를 vincent6 어셈블리 코드로 구현하였다. 최종적으로 구현된 코드는 ITU-T 의 test vector 에 대 해 bit exact 한 결과를 보이며 34 MCPS (Million Cycles Per Second)의 계산량을 가지며 사용 메모리크기는 데이터 메모리가 약 9KByte, 프로그램 메모리가 약 57 KByte 이다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.52-55
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• 1999

다중 전송율(16, 24, 32, 40kbps)을 제공하는 G.726 부호화기는 ADPCM (Adaptive Differential Pulse Code Modulation) 부호화법을 사용한다. 본논문에서는 G.726 ADPCM 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00A 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.726에 대한 양방향 평가는 Codec Loopback test을 통해 수행되었으며, W-T에서 제공한 테스트 절차에 따라 평가되었다. 본 논문에서 구현된 G.726 부호화기는 평균 11 MIPS의 계산 속도를 갖고, 프로그램 메모리 크기는 2.8K Words이고, 데이터 메모리 크기는 550 Words 를 필요로 하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2017.06a
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• pp.133-134
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• 2017

본 논문에서는 차세대 방송 표준 ATSC (Advanced Television Systems Committee) 3.0 기반 LDM (Layered Division Multiplexing) 및 스케일러블 비디오 코덱 (Scalable Video Codec) 을 활용한 지상파 방송시스템 기술을 살펴보고 그 성능을 비교 분석한다. 코어 레이어 (Core Layer)와 인핸스드 레이어 (Enhanced Layer)로 구성된 LDM 기반 PLP (Physical Layer Pipe)에, 스케일러블 비디오 코딩이 적용된 베이스 레이어 (Base Layer)와 인핸스먼트 레이어 (Enhancement Layer) 스트림을 각각 전송함으로써 하나의 RF 채널에 두 개 이상의 서비스를 전달할 경우 채널 효율을 극대화 할 수 있다. 본 논문에서는 이동 및 고정용 서비스, 즉 두 개의 서비스를 전송할 때 제안된 LDM 및 스케일러블 비디오 코덱을 사용한 기술과 TDM (Time Division Multiplexing) 및 Simulcast 를 적용한 기술과의 성능 비교를 통해 제안된 기술의 우수성을 검증하고자 한다.