• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.530-533
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• 1999

During the last decade, high quality digital audio has essentially replaced analog audio. During this period, digital audio have applied many application areas of the info-industry. These applications have created a demand for high quality digital audio. In audio compression, the methods using human auditory nervous properties are used and introduced from psychoacoustical model utilized perceptual audio coding unable to code above the limitation of human perception. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustical models to exploit efficiently masking characteristics of the human receiver. In this paper, the designed MDCT/IMBCT as a standard of current MPEG is implemented onto FPGA.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.329-332
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• 2005

In this paper, we describe implementation of personal media gateway (PMG) for applications over BcN networks. PMG is a TV based set-top terminal, which enables transmission of Full D1 high quality video and audio at the speed of maximum 2Mbps. It supports SIP protocol and QoS for the BcN networks. The hardware of the PMG consists of host module, audio/video codec processing module, DTMF module, and remote control I/O module. H.263 and MPEG4 software are implemented in DSP as codec for hi-directional communication and streaming, respectively. G.711 and Ogg-Vorbis are implemented as audio codec. We examined the quality of video using the Video Quality Test Equpment, which was developed by KT Convergence Lab. The experimental results show the video quality of MOS 4.1 and audio quality of MOS 4.3. We expect that PMG will be prospective business models, and create new customer value.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• International journal of advanced smart convergence
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• v.6 no.2
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• pp.51-58
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• 2017

Recently, rapid advances of Ethernet and IP technology have brought many changes in the sound industry. In addition, due to AoIP-based audio transmission technology, various problems of the acoustic system (sound quality deterioration due to long distance transmission, complicated wiring) have improved dramatically. However, when many distributed audio systems are connected with AoIP equipment, if there is a problem in the equipment, it is impossible to operate the connected system. AoIP equipment only can transmit audio signals but cannot adjust the system for acoustic environment. In this paper, AoIP equipment is to be installed with sound equipment on a one-to-one basis, so that various existing problems can be solved and adjustment of sound quality (reverberation, echo, delay and EQ) can be possible by AoIP-based OAC (On-site Audio Center) with built-in DSP function. As a result, uncompressed real-time transmission by distributed transmission/receipt module in OAC (On-site Audio Center) and high quality sound by adjustment of sound quality with built-in DSP can be acquired. It is expected that OAC based sound system will be the industry standard in ubiquitous environment.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.12
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• pp.716-722
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• 2005

In this paper, we propose the audio watermarking algorithm based on the critical band of HAS(human auditory system) without audibly affecting the quality of the watermarked audio and implement the detecting algorithm on the BSS(broadcast synchronizing system) for testing the proposed algorithm. According to the audio quality test, the SNR(signal to noise ratio) of the watermarked audio objectively is 66dB above. In the robustness test, the proposed algorithm can detect the watermark more than $90\%$ from various compression(MP3, AAC), A/D and D/A conversions, sampling rate conversions and especially asynchronizing attacks. The BSS automatically switches the programs between the key station and the local station in broadcasting system. The result of reliability test of implemented system by using the real broadcasting audio has no false positive error during 30 days. Because of detecting once processing per 0.5 second, we can judge that the false positive error does not occur.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.5 s.311
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• pp.48-56
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• 2006

In this paper, we proposed the high quality audio watermarking algorithm using MDCT/IMDCT (Modified DCT/Inverse Modified DCT) with psychoacoustic model. Generally, a digital audio watermark is embedding the frequency domain after frequency transform of the digital audio data but the digital audio quality is affected by watermarking. In our scheme, the digital audio data is spread with PN((Pseudo Noise) code and then audio watermark is embedded in MDCT processing that refers psychoacoustic model. In MDCT processing, according to the shape of filter bank output, the block switching selects a window sequence that has 256, 1,024 or 2,048 points interval for high quality audio. The author confirm that when watermark weight ${\alpha}$ is 2.5 below, the detection ratio of watermark is a satisfied to SDMI's(Secure Digital Music Initiative) recommendation 50% above and SM is $50{\sim}68dB$ range with mainly 4 kind of attacks(Compression, Cropping, FFT and Echo).

• 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.

• The Journal of the Acoustical Society of Korea
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• v.32 no.3
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• pp.214-226
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• 2013

Assessing the quality of audio signals is an important consideration in making high quality sounds and various methods have been developed. This paper provides a general framework of sound quality and a technical overview of the international standard methods which are described in ITU-T, ITU-R, IEC and ANSI Recommendations in the speech intelligibility, speech quality, and audio quality areas. In addition, some recent findings and future works are included.

• Broadcasting and Media Magazine
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• v.4 no.1
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• pp.62-79
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• 1999

In this survey paper the emerging MPEG-4 audio technology is discribed In the previous MPEG-1 and the MPEG-4 audio words, only the natural audio and the speech coding techniques were the standadization objects But in the MPEG-4 audio standadization, not only the natural audio and the speech coding, but also the structured audio and the synthetic speech techniques are inclued, The purpose of this expansion can be summarized as the preparation for the versatile high-quality multimedia services supposed emerge in the 21st century.