• Proceedings of the KIEE Conference
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• 2005.05a
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• pp.170-172
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• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, use 1 bit representation with a sampling frequency of 2.8224 MHz (64 $\times$ 44.1 kHz). For multi-rate PCM (Pulse Code Modulation) input like as 48/96/192 kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital filter structure composed of sample-rate converter and interpolation filter for the DSD encoder with multi-rate (48/96/192 kHz) PCM input. without a external sample-rate converter.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.375-376
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• 2006

In this paper, we propose a wavelet-based audio watermarking algorithm to be robust against capturing attack. Commercial capturing tools enable us to obtain audio contents without noticeable degradation in audio quality, and it is possible to be a source of illegal distribution. By adjusting mean values of the lowest subband in audio, the proposed method can survive after capturing attack including sampling rate conversion, random cropping and compression. By applying a simple human auditory model, the inaudibility of the watermark is achieved, and detection probability is improved based on the difference information. This is confirmed by experimental results.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.358-360
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• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, uses 1 bit representation with a sampling frequency of 2.8224MHz (64${\times}$44.1kHz). For multi-rate PCM (Pulse Code Modulation) input such as 8${\sim}$192kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital mter structure composed of sample-rate converter and interpolaton filter for the DSD encoder with multi-rate (8${\sim}$192kHz) PCM input, without a external sample-rate converter.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.150-152
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• 2004

Conventional CDDA (Compact Disc Digital Audio) system has limitations come from sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. So, new medium is developed for high-resolution audio recording, like as DVD-audio etc. But CDDA is a widely used medium for high fidelity audio yet, because new medium has complexity and difficulty in manufacturing system. In this paper, we design a new encoding system for high-resolution audio signal. The system is backward compatible with conventional CDDA. By evaluation for encoding and decoding process, we describe practicability of our proposal system.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.5
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• pp.327-329
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• 2005

Conventional CDDA(Compact Disc Digital Audio) system has limitation which means that bandwidth and resolution of the sign릴 are determined by the sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. Though, new medium such as DVD-audio is developed for high-resolution audio recording, it has high complexity and difficulty in manufacturing process. So, CDDA is a widely used medium for high fidelity audio yet. In this paper, we design a new encoding system for high-resolution audio signal which has backward compatible with conventional CDDA. By evaluating for the encoding and decoding process. we verify the availability of our proposed system.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.54-60
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• 2004

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.127-134
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• 2009

In spite of controversial debates on aesthetic issues of computer-generated electronic music, rapid advancement of music technologies in the past decade have resulted proliferation of using virtual software synthesizers and samplers in music composition. Computer-based music production platform has become not only a norm among some of contemporary music composers but also vital apparatus for their compositional process. There are two imperative parts of this compositional process involving sampling in computer-based music production, which are commercially available sample libraries that include pre-recorded audio samples, and music production software that processes them. The purpose of this study is to investigate the process and effectiveness of reconstructive compositional process utilizing distinctive features of sampling on computer music production software. This study addresses issues such as: the definition of audio sampling, how sampling is incorporated in compositional process, and what features of music production software are particularly effective in various musical expressions. The result of this study will hopefully accommodate and fulfill the needs of electronic and acoustic musicians' creativeness.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.187-190
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• 2002

본 논문에서는 지금까지 소개된 44.1KHz compact disc (CD)에서 48KHz digital audio tape (DAT)로의 표본화율 변환기법들에 대해서 가청 주파수 대역에서 100dB 이상의 dynamic range와 ±5x10­4dB 이하의 리플 크기를 유지할 수 있도록 각 기법들을 재설계하였으며, 메모리 요구량 및 계산량에 대해서 살펴보고자한다.