• MALSORI
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• no.53
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• pp.157-169
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• 2005

This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 2001.11a
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• pp.464-468
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• 2001

In this paper, we propose a robust watermarking technique that accepts time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC, WMA. The technique is developed based on digital filtering. Being designed according to critical band of HAS (human auditory system), the digital filters nearly affect audio quality. Furthermore, before implementing digital filtering, wavelet transform decomposes the audio signal into several signals that is composed of specific frequencies. Designed digital filters scan the decomposed signal. The designed digital filter, band-stop filter, distorts and eliminates specific frequencies of audio signals. Watermarking detection can be accomplished by FFT (Fast Fourier Transform). Firstly, segments of audio signal are transformed by FFT. Then, the obtained amplitude spectrum by FFT is summed repeatedly. Finally the watermark detector can find filters used to watermark encoding based on eliminating frequencies. The suggested technique can embed 4bits/s in a robust manner.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

• ETRI Journal
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• v.24 no.3
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• pp.181-189
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• 2002

Digital watermark technology is now drawing attention as a new method of protecting digital content from unauthorized copying. This paper presents a novel audio watermarking algorithm to protect against unauthorized copying of digital audio. The proposed watermarking scheme includes a psychoacoustic model of MPEG audio coding to ensure that the watermarking does not affect the quality of the original sound. After embedding the watermark, our scheme extracts copyright information without access to the original signal by using a whitening procedure for linear prediction filtering before correlation. Experimental results show that our watermarking scheme is robust against common signal processing attacks and it introduces no audible distortion after watermark insertion.

• 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.

• ETRI Journal
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• v.33 no.6
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• pp.945-948
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• 2011

Object-based audio coding can provide new music applications with interactivity. To efficiently compress a lot of target audio objects, a subband-based parametric coding scheme has been adopted for MPEG spatial audio object coding. In this letter, the time-frequency (T/F) subband analysis structure is investigated. A reconfigured T/F structure is also proposed to enhance the generating performance of sound scenes such as 'karaoke' and 'solo' play in interactive music scenarios. From the experimental results, it was confirmed that the proposed scheme remarkably improves the SNR and sound quality.

• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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• pp.174-182
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• 2011

Conventional home audio system (e.g. AV Receiver, CD Player etc) has a various functionality of audio play, channel mixing, but the remote controller of these audio players is too complex, difficult for user to manage them effectively. Users want to use these functionalities with more easy, comprehensible way. In this study, "intelligent real-sound presentation technology" that support high quality, realistic audio and the "design of complex information and controller of real sound using intelligent real sound play and control interface" will be introduced. So user can actively, realistically enjoy and play real sound based on user's preference, emotion and circumstance, instead of user's passive service.

• Journal of Information Processing Systems
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• v.15 no.2
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• pp.271-287
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• 2019

The synchronization scheme based on moving average is robust and suitable for the same rule to be adopted in embedding watermark and synchronization code, but the imperceptibility and search efficiency is seldom reported. The study aims to improve the original scheme for robust audio watermarking. Firstly, the survival of the algorithm from desynchronization attacks is improved. Secondly, the scheme is improved in inaudibility. Objective difference grade (ODG) of the marked audio is significantly changed. Thirdly, the imperceptibility of the scheme is analyzed and the derived result is close to experimental result. Fourthly, the selection of parameters is optimized based on experimental data. Fifthly, the search efficiency of the scheme is compared with those of other synchronization code schemes. The experimental results show that the proposed watermarking scheme allows the high audio quality and is robust to common attacks such as additive white Gaussian noise, requantization, resampling, low-pass filtering, random cropping, MP3 compression, jitter attack, and time scale modification. Moreover, the algorithm has the high search efficiency and low false alarm rate.

• Journal of Korea Multimedia Society
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• v.16 no.10
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• pp.1171-1179
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• 2013

Interactive audio service provide with audio generating and editing functionality according to user's preference. A spatial audio object coding (SAOC) scheme is audio coding technology that can support the interactive audio service with relatively low bit-rate. However, when the SAOC scheme remove the specific one object such as vocal object signal for Karaoke mode, the scheme support poor quality because the removed vocal object remain in the SAOC-decoded background music. Thus, we propose a new SAOC vocal harmonic extranction and elimination technique to improve the background music quality in the Karaoke service. Namely, utilizing the harmonic information of the vocal object, we removed the harmonics of the vocal object remaining in the background music. As harmonic parameters, we utilize the pitch, MVF(maximum voiced frequency), and harmonic amplitude. To evaluate the performance of the proposed scheme, we perform the objective and subjective evaluation. As our experimental results, we can confirm that the background music quality is improved by the proposed scheme comparing with the SAOC scheme.