• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.18-25
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• 2008

In this paper, new lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder uses Huffman coding utilizing the statistical characteristics of spectral coefficients, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, new lossless coding scheme with better performance is proposed that consists of bit-plane transform and run-length coding. In the proposed scheme, the spectral coefficients are first transformed by bit-plane into 1-D bit-stream with better correlative properties, which is then coded intorun-length and is finally Huffman coded. In addition, the coding performance is further increased by applying the proposed bit-plane coding selectively to each group, after the entire frequency is divided into 3 groups. The performance of proposed coding scheme is measured in terms of theoretical number of bits based on the entropy, and shows at most 6% enhancement compared to that of conventional lossless coder used in AAC audio codec.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.137-140
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• 2004

낮은 대역폭에서 다채널 다객체 오디오 신호를 전송하기위해 새롭게 제안된 Spatial Audio Coding 기술은 멀티채널 오디오 신호를 다운믹싱하고 나머지 채널은 음향공간상의 위치정보를 나타내는 파라미터들로 압축하여 표현하는 파라메트릭 압축 방식이다. 본 논문에서는 Spatial Audio Coding 기술중의 하나인 BCC 기술을 이용하여 스테레오 오디오 코덱을 구현하고, 주관듣기평가 실험을 통하여 AAC와 비슷한 성능을 나타내면서도 높은 압축율을 얻을 수 있음을 확인하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.10
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• pp.1914-1919
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• 2006

Video transfer problem on mobile is transfer speed and controls. Compression technique is needed to transfer videos and H.263 codec is used for compression, effectively controls camera on remote places, increased the real time connecting users. In this paper, we could solve the problem that use existent RF, and could transfer the most suitable image and audio.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.11
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• pp.13-20
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• 2005

For Modern Consumer and Communication Elecronic Devices, Always Error Protecting HW and SW is used. The Core is RS(Reed Solomon) Codec in Galois Field GF($2^8$). Here New 2 to 3 Symbol RS Decoder Design and Encoder design Method using Normalized error position Value is described. Examples are given to show the methods are working well.

• ETRI Journal
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• v.31 no.6
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• pp.732-740
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• 2009

This paper describes the implementation of a digital audio effect system-on-a-chip (SoC), which integrates an embedded digital signal processor (DSP) core, audio codec intellectual property, a number of peripheral blocks, and various audio effect algorithms. The audio effect SoC is developed using a software and hardware co-design method. In the design of the SoC, the embedded DSP and some dedicated hardware blocks are developed as a hardware design, while the audio effect algorithms are realized using a software centric method. Most of the audio effect algorithms are implemented using a C code with primitive functions that run on the embedded DSP, while the equalization effect, which requires a large amount of computation, is implemented using a dedicated hardware block with high flexibility. For the optimized implementation of audio effects, we exploit the primitive functions of the embedded DSP compiler, which is a very efficient way to reduce the code size and computation. The audio effect SoC was fabricated using a 0.18 ${\mu}m$ CMOS process and evaluated successfully on a real-time test board.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.662-675
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• 2018

Compression history detection plays an important role in digital multimedia forensics. Most existing works, however, mainly focus on digital image and video. Additionally, the existed audio compression detection algorithms aim to detect the trace of double compression. In real forgery scenario, multiple compression is more likely to happen. In this paper, we proposed a detection algorithm to reveal the compression history for MP3 audio. The statistics of the scale factor and Huffman table index which are the parameters of MP3 codec have been extracted as the detecting features. The experimental results have shown that the proposed method can effectively identify whether the testing audio has been previously treated with single/double/triple compression.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.345-346
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• 2007

In this paper, we propose a Pyramid VQ(PVQ) to quantize the transform coefficients of TCX module for the music improvement of AMR-WB+ codec. The proposed PVQ is compared to the $RE_8$ Lattice VQ used in the AHR-WB+ standard codec, demonstrating improvement 4% and 5.7%, respectively, in Mean Squared Error(MSE) and 3.3% and 4.7%, respectively, in Perceptual Evaluation of Audio Quality(PEAQ) by 8-dimensional and 16-dimensional Pyramid VQ.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.357-368
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• 2016

In DAB+, HE-AAC v2 codec is applied for the fixed rate transmission of audio stream. In case that xHE-AAC codec including USAC, a more efficiency is expected when the variable frame is used in a given same bandwidth compared to the fixed frame transmission. For this to be realized, audio streams need to be multiplexed in a sub-channel before transmission, then a method is required to identify the border of each audio frames. In this paper, the toggled sync byte and additional identification field being sequentially placed between AU borders are proposed in order to deal with the AU border identification. In addition, the Reed-Solomon based error correction code which is compliant to DAB+ is proposed.

• ETRI Journal
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• v.22 no.1
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• pp.20-29
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• 2000

In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.