• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.328-338
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• 2005

Technology for compressing low-bitrate multichannel audio coding is being standardized owing to the increasing need of consumer for multichannel audio contents. In this paper we propose the sound source location cue coding (SSLCC) for extremely compressing multichannel audio to be suitable at the narrow bandwidth transmission environment. To improve the compression capability of the conventional binaural cue coding(BCC), the SSLCC adopts the virtual source location information (VSLI) as a spatial cue parameter, a symmetric uniform quantizer, and Huffman coder. The objective and subjective assessment results show that the SSLCC provides lower bitrate and better audio quality than conventional BCC method.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.74-78
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• 2004

As analog audio systems are changing to digital systems, the DAB (Digital Audio Broadcasting) is expected to provide CD quality audio, various data services with interactiveness and excellent mobile reception ability. The DMB (Digital Multimedia Broadcasting), as more advanced successor of the DAB, adds video capability on the audio and data services. The DAB system assures high quality audio services even when the reception is through portable and mobile receivers. In this paper, USB-DAB receiver and PCI-DMB receiver are designed and implemented. The DAB receiver and the DMB receiver incorporate with PC to make use of computational power and application software of Pc. This enables the developed system to be more flexible and to meet various applications easier.

• JSTS:Journal of Semiconductor Technology and Science
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• v.5 no.4
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• pp.249-254
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• 2005

This paper presents a system-on-a-chip (SOC) design for digital TV. The single LSI incorporates almost all essential parts such as CPU, ISO/IEC 11172/13818 system/audio/video decoders, a video post-processor, a graphics/OSD processor and a display processor. It has analog IP's inside such as video DACs, an audio PLL, and a system PLL to reduce the system-level implementation cost. Descramblers and Smart Card interface are included to support widely used conditional access systems. The video decoder can decode two video streams simultaneously. The DSP-based audio decoder can process various audio coding specifications. The functional blocks for video quality enhancement also form outstanding features of this SoC. The SoC supports world-wide major DTV services including ATSC, ARIB, DVB, and DIRECTV.

• 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.

• Convergence Security Journal
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• v.6 no.4
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• pp.21-27
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• 2006

With the introduction of sound source sharing over the high speed internet and portable digital audio, the digitalization of sound source has been rapidly expanded and the sales and distribution of sound sources of the former offline markets are stagnant. Also, the problem of infringement of copyright is being issued seriously through illegal reproduction and distribution of digitalized sound sources. To solve these problems, the DRM technology for protecting contents and copyrights in portable digital audio device began to be introduced. However, since the existing DRM was designed based on the fast processing CPU and network environment, there were many problems in directly applying to the devices with small screen resolution, low processing speed and network function such as digital portable audio devices which the contents are downloadable through the PC. In this study, the DRM structural model which maintains similar security level as PC environment in the limited hardware conditions such as portable digital audio devices is proposed and analyzed. The proposed model chose portable digital audio exclusive device as a target platform which showed much better result in the aspect of security and usability compared to the DRM structure of exiting portable digital audio device.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.505-506
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• 2004

이어폰과 헤드폰의 성능 향상을 위해서는 특성에 대한 정확한 측정과 평가가 요구된다. 이어폰과 헤드폰은 room 과 같은 acoustic channel 을 거치지 않고 청취자의 귀에 직접 소리가 전달되며 ear canal 특성이 포함되기 때문에 스피커와는 다른 기준이 필요하다. 그러나 사람 귀의 canal 특성은 개인에 따른 편차가 심하여 정확한 측정 및 성능 평가에 어려움이 따른다. 본 논문에서는 이어폰과 헤드폰의 특성을 측정하는 적절한 방법을 고찰하고 측정된 데이터를 이용하여 음질 성능을 평가할 수 있는 객관적인 방법을 제시하고자 한다.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.60 no.1
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• pp.19-26
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• 2011

In this paper, we designed Interpolation FIR(Finite Impulse Response) filter and 1-bit SDM(Sigma- Delta Modulator) for small digital audio speaker, which has low power consumption and high output characteristics. In order to achieve high linearity and low distortion performance of the systems, we adopt Type I Chevychev FIR filter which has equiripple characteristics in the pass band and proposed high efficient FIR filter structure. SDM is the most efficient modulation technique among the noise shaping techniques. In this paper, we implemented SDM using CIFB(Cascade of Intergrators, Feed-Back) which is generally used in DAC of small digital audio speakers. The proposed SDM structure can achieve high SNR, high-efficiency characteristics and low power consumption in mobile devices. Also considering manufacture of SoC(System on Chip), we performed simulation with Matlab and Verilog HDL to obtain optimal number of operational bits and verified a good experimental results.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.186-193
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• 2005

The Terrestrial Digital Multimedia Broadcasting (T-DMB) system supplies high quality audio comparable with VCD in 7 inch display and high quality audio comparable CD at the mobile reception environment T-DMB will launch commercial service at the middle of 2005. However the bandwidth for audio data and the number of channels are restricted to 128 kbps and 2 respectively in the current T-DMB standard because of the limitation of available bandwidth for multimedia data. This Paper Proposes a novel media processor structure for providing multi-channel audio contents oyer T-DMB system allowing backward compatibility with the legacy T-DMB receiver. Furthermore. we also Propose an adaptive receiver structure to supply optimal audio contents on various speaker configuration in T-DMB receiver. To provide multi-channel audio contents allowing backward comaptilbity with the legacy T-DMB receiver, the additional data for multi-channel audio are defined as a dependent stream of main audio stream. The OD strucure for control an additional multi-channel audio elementary stream is proposed without changing the BIFS of the legacy T-DMB system.

• Journal of Digital Contents Society
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• v.18 no.5
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• pp.811-820
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• 2017

For content-based audio retrieval which is one of main functions in audio service, the techniques for extracting fingerprints from the audio source, storing and indexing them in a database are widely used. However, if the fingerprints of new audio sources are continually inserted into the database, there is a problem that space efficiency as well as audio retrieval performance are gradually deteriorated. Therefore, there is a need for techniques to support efficient expansion of audio database without periodic reorganization of the database that would increase the system operation cost. In this paper, we design a content-based audio retrieval system that solves this problem by using MapReduce and NoSQL database in a cluster computing environment based on the Shazam's fingerprinting algorithm, and evaluate its performance through a detailed set of experiments using real world audio data.