• The Journal of the Acoustical Society of Korea
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• v.30 no.6
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• pp.303-307
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• 2011

This paper describes an audio watermarking using Quantization Index Modulation (QIM) on significant peaks in frequency domain. The audio signal is broken up into L samples length frames with non-overlapping and rectangular window. The zero-crossing rate of each frame is calculated for decision whether it is proper to be watermarked or not. If the frame is legitimate, frequency magnitude response is computed by discrete Fourier transform. For the QIM, we set the quantization step size based on maximum value of frequency magnitude response and select n significant peaks with w samples around them in frequency domain, totally $n{\times}(w+1)$ samples. Finally, watermark embedding is performed. Decoder extract watermarks based on Euclidean distance, that is a blind detection. The proposed method is robust against many attacks of watermark benchmark.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1296-1299
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• 2002

We implemented a CD signal processor operated on a CAV 48-speed CD-ROM drive into a VLSI. The CD signal processor is a mixed mode monolithic IC including servo-processor, data recovery, data-processor, and I-bit DAC. For servo signal processing, we included a DSP core, while, for CAV mode playback, we adopted a PLL with a wide recovery range. Data processor (DP) was designed to meet the yellow book specification.［2］So, the DP block consists of EFM demodulator, C1/C2 ECC block, audio processor and a block transferring data to an ATAPI chip. A modified Euclid's algorithm was used as a key equation solver for the ECC block To achieve the high-speed decoding, the RS decoder is operated by a pipelined method. Audio playability is increased by playing a CD-DA disc at the speed of 12X or 16X. For this, subcode sync and data are processed in the same way as main data processing. The overall performance of IC is verified by measuring a transfer rate from the innermost area of disc to the outermost area. At 48-speed, the operating frequency is 210 ㎒, and this chip is fabricated by 0.35 um STD90 cell library of Samsung Electronics.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.4
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• pp.355-355
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• 2002

In this paper, we designed a FPU(floating point unit) that it is very important and requires of high density when digital audio is designed. Almost audio system must support the multi-channel and required for high quality. A floating point arithmetic function in MPEG-2 AAC that implemented by hardware is able to realtime decoding when DSP realization. The reason is that MPEG-2 AAC is compatible to the Audio field of MPEG-4 and afterwards. We designed a FPU by hardware to increase the speed of a floating point unit with much calculation part in the MPEG-2 AAC Decoder. A FPU is composed of a multiplier and an adder. A multiplier used the Radix-4 Booth algorithm and an adder adopted 1's complement method for speed up. A form of a floating point unit has 8bit of exponent part and 24bit of mantissa. It's compatible with the IEEE single precision format and adopted a pipeline architecture to increase the speed of a processor. All of sub blocks are based on ISO/IEC 13818-7 standard. The algorithm is tested by C language and the design does by use of VHDL(VHSIC Hardware Description Language). The maximum operation speed is 23.2MHz and the stable operation speed is 19MHz.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.7
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• pp.823-832
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• 2016

This paper presents a generation method of test streams for verifying conformance of an UHD broadcasting receiver including decoders for video and audio as well as parsers for PSIP and closed caption data. The proposed test streams for video/audio signals can evaluate conformance of HEVC, AC-3 and DTS-HD standards. Especially, test streams for HEVC video compression standard can be used for testing syntax compliance and error resilience for a HEVC decoder. Moreover, the proposed test streams for system/program and closed caption can be applied for verifying parsers for PSIP and CEA-708 standards.

• Journal of Broadcast Engineering
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• v.24 no.3
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• pp.387-394
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• 2019

In this study, a new method of audio high-band coding based on autoencoder with side information is proposed. The proposed method operates in the MDCT domain, and improves the performance by using additional side information consisting of the previous and current low bands, which is different from the conventional autoencoder that only inputs information to be encoded. Moreover, the side information in a time-frequency domain enables the high-band coder to utilize temporal characteristics of the signal. In the proposed method, the encoder transmits a 4-dimensional latent vector computed by the autoencoder and a gain variable using 12 bits for each frame. The decoder reconstructs the high band by applying the decoded low bands in the previous and current frames and the transmitted information to the autoencoder. Subjective evaluation confirms that the proposed method provides equivalent performance to the SBR at approximately half the bit rate of the SBR.

• Journal of Broadcast Engineering
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• v.28 no.2
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• pp.213-229
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• 2023

In this paper, we introduce a spatial sound rendering system that provides 6DoF spatial sound in real time in response to the movement of a listener located in a virtual environment. This system was implemented using MPEG-I AEP as a development environment for the CfP response of MPEG-I Immersive Audio and consists of an encoder and a renderer including a decoder. The encoder serves to offline encode metadata such as the spatial audio parameters of the virtual space scene included in EIF and the directivity information of the sound source provided in the SOFA file and deliver them to the bitstream. The renderer receives the transmitted bitstream and performs 6DoF spatial sound rendering in real time according to the position of the listener. The main spatial sound processing technologies applied to the rendering system include sound source effect and obstacle effect, and other ones for the system processing include Doppler effect, sound field effect and etc. The results of self-subjective evaluation of the developed system are introduced.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10b
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• pp.221-223
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• 1999

소프트웨어로 MPEG-1 디코더의 구현이 가능하다. 소프트웨어 MPEG-1 디코더의 문제 중 하나는 MPEG-1 압축 알고리즘의 특징상 각각의 영상들이 서로 다른 압축율로 압축되기 때문에 재생시에 디코더에 걸리는 부하가 매우 불규칙적이라는 점이다. 이 문제는 MPEG-1 디코더를 보다 작은 실행 단위인 쓰래드로 나누어 처리함으로써 효율적으로 해결할 수 있다. 이때 독립적인 실행 흐름을 가지는 쓰래드들간의 데이터 전달을 위하여 공유버퍼를 사용하게 된다. 본 논문에서는 다중 쓰래드로 구성된 소프트웨어 MPEG-1 디코더에서 쓰래드들 간의 데이터 전달에 사용되는 공유 버퍼를 이용하여 오디오와 비디오 스트림의 동기화를 효과적으로 수행하는 기법을 소개한다.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.174-177
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• 2000

본 논문에서는 높은 압축률과 고음질을 제공하는 MPEG-1 Layer Ⅲ 오디오 디코더를 고정소수점 DSP인 TMS320C6201을 이용하여 실시간으로 동작하도록 구현하였다. ISO/IEC에서 제공하는 부동소수점 C 프로그램을 음질의 손실 없이 고정소수점 연산으로 변환하었고 실시간 동작을 위하여 최적화 작업을 수행하였다. 연산의 정확성을 높이기 위해서 Descaling 모듈에 중점을 두어 부동소수점 연산을 고정소수점 연산으로 변환하였고 IMDCT 모듈과 Synthesis Polyphase Filter Bank 모듈에 대해 고속 알고리즘을 적용하여 연산량과 프로그램 크기를 크게 줄일 수 있었다. 구현된 디코더는 TMS320C6201 DSP가 수행할 수 있는 최대 연산량의 26%만으로 실시간 동작이 가능하였고 부동소수점 연산 결과와 고정소수점 연산 결과를 비교하여 60 dB 이상의 높은 SNR을 가짐을 확인하였다. 또한 사운드 입출력과 호스트 통신을 통하여 EVM 보드에서 실시간으로 동작함을 확인하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.181-184
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• 1999

본 논문에서는 MPEG-1 계층 II와 MPEG-2 계층 II LSF 오디오 복호기를 OAK DSP Core를 이용하여 실시간 응용이 가능하도록 구현하였다. Ungrouping시 이용되는 테이블을 효율적으로 사용하였으며 합성필터부의 RAM과 ROM의 크기 그리고 각 부분의 연산에 필요한 연산량을 최적화하기 위하여 알고리듬을 효율적으로 적용하였고 불필요한 연산 부분을 제거하거나 최적화 하였다.