• Journal of the Korea Convergence Society
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• v.7 no.3
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• pp.77-85
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• 2016

We propose a digital signal processing technic that can compensate the non-linearity inherent in audio amplifiers, and present the result of the simulation. The inherent non-linearity of the audio power amplifier arising from analog devices is compensated via a digital signal processing technic consisting of indirect learning architecture and an adaptive filter. The simulation results show that the compensator can be realized using a third-order polynomial and compensates odd-order non-linearity efficiently. The even-oder non-linearity is mainly due to the dc offset at the output, which is difficult to eliminate with the proposed method. Care must be taken in designing the bias circuit to avoid the DC offset at the output. The proposed technic has significance in that digital signal processing technic can compensate for the impairment that is an inherent characteristic of an analog system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.615-618
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• 2007

It has able to express digital video/audio data from analog and to broadcast it via improvement of video/audio compression technology and publishing standard of MPEG-2 System. Nowaday many System Operators are provide regular digital broadcasting program to customer with their own access program. To provide access program, two process needs that are creating broadcasting information and remultiplexing it with video/audio data, and this solution is providing with high-cost system only. For this reason, digital access program bas week point to product. In this paper, we designed and implemented Generating PSIP Information System to product digital access program which generate PSIP information via receiving broadcasting information from user, and map PSIP information directly to video/audio data.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2101-2102
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• 2006

In this paper, a new digital audio watermarking algorithm is presented. The proposed algorithm embeds watermark into audio signal based on human auditory system (HAS). This algorithm is a blind audio watermarking method, which does not require any prior information during watermark extraction process. This algorithm finds watermarking position using time-domain masking effect. First we insert the watermark into wavelet domain, and then we use a back-propagation neural network (BPN) to learn the characteristics of relationship between the watermark and the watermarked audio. Due to the teaming and adaptive capabilities of the BPN, the false recovery of the watermark can be greatly reduced by the trained BPN. Experimental results show that the proposed method has good inaudibility and high robustness to common audio processing attacks.

• Journal of KSNVE
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• v.9 no.5
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• pp.991-997
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• 1999

It is getting higher the demand on a good sound in a car. Like some other sensory qualities, achieving a good sound is still treated as a black-box followed by the individual taste of tuning engineers. This study aims to define the attributes of a good car audio sound and the assessment factors affecting the sound quality. With a variety of surveys on professional listening groups and car audio installers, 26 most frequently used vocabularies are selected as typical presentations of car audio sound quality. After repetitions of listening tests with deliberately selected test CD, pre-recorded 15 different digital car-recordings, headphones replay system, and three different listening groups, it is proved that there are at least 5 different assessment factors which significantly affect the car audio sound quality. They are high fidelity, spectral balance, tonal accuracy, sound stage and acoustic fault.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.108-115
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• 1999

In this study we suggested a digital filter method to enhance car audio sound quality against the sound distortion due to cabin's acoustic characteristics and car driving noises. The digital filters designed were based on the characteristics on car driving noises and cabin acoustic characteristics. Car driving noises were analyzed by two ways; one is an objective method, octave band frequency analysis method. The other is a subjective method; sensory evaluation method, NCB method. On these results, seven sets of modified coefficients of eleven band digital filters were obtained. To find optimum audio sound quality among nine sound samples filtered by designing seven types of digital filters, which were mixed car driving noises at 100km/h, subjective evaluation method was used, paired comparison method; Scheffe' seven point method.

• Journal of Korea Multimedia Society
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• v.25 no.2
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• pp.166-175
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• 2022

Digital transformation is driving so many changes in daily life and industry. The automobile industry is in a similar situation. In some cases, element techniques in areas called metabuses are also being adopted, such as 3D animated digital cockpit, around view, and voice AI, etc. Through the growth of the mobile market, the norm of human-computer interaction (HCI) has been evolving from keyboard-mouse interaction to touch screen. The core area was the graphical user interface (GUI), and recently, the audio user interface (AUI) has partially replaced the GUI. Since it is easy to access and intuitive to the user, it is quickly becoming a common area of the in-vehicle experience (IVE), especially. The benefits of a AUI are freeing the driver's eyes and hands, using fewer screens, lower interaction costs, more emotional and personal, effective for people with low vision. Nevertheless, when and where to apply a GUI or AUI are actually different approaches because some information is easier to process as we see it. In other cases, there is potential that AUI is more suitable. This is a study on a proposal to actively apply a AUI in the near future based on the context of various scenes occurring to improve IVE.

• Broadcasting and Media Magazine
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• v.21 no.3
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• pp.36-45
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• 2016

본 고에서는 디지털 라디오 방송 수신 기술 규격인 DAB(Digital Audio Broadcasting), DAB+ 기술과 DRM(Digital Radio Mondiale), DRM+ 그리고 최신 오디오 압축 기술인 USAC(Unified Speed and Audio Coding) 코덱 기술이 적용된 디지털라디오 수신 기술을 설명하고자 한다. 이를 위해 기존의 디지털라디오 규격과 오디오 코덱 기술을 살펴보고 오디오 서비스의 품질을 높이고 주파수 효율을 증대시킬 수 있는 USAC 기술을 살펴본다. 그리고 이를 적용할 수 있는 디지털라디오 수신 기술을 제시한다. 향후 DAB+기반의 디지털라디오 수신기에 USAC 적용 가능성을 제시하며 마무리 짓는다.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.716-719
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• 1999

In this Paper, we Present a method for embedding digital watermarks into digital audio signals. The watermarking must be imperceptible and should be robust to attacks, such as filtering and compression etc. In our method, we adaptively embedded the watermarks changing the scale factor using the spread spectrum and MPEG-2 AAC psychoacoustic model.

• Journal of KIISE:Databases
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• v.31 no.4
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• pp.341-351
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• 2004

With the popularity of digital audio contents, querying and retrieving audio contents efficiently from database has become essential. In this paper, we propose a new index scheme for retrieving audio contents efficiently using audio portions that have been queried frequently. This scheme is based on the observation that users have a tendency to memorize and query a small number of audio portions. Detecting and indexing such portions enables fast retrieval and shows better performance than sequential search-based audio retrieval. Moreover, this scheme is independent of underlying retrieval system, which means this scheme can work together with any other audio retrieval system. We have implemented a prototype system and showed its performance gain through experiments.

• Journal of Digital Contents Society
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• v.18 no.1
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• pp.183-190
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• 2017

This paper proposes an audio event detection method using Deep Neural Networks (DNN). The proposed method applies Feed Forward Neural Network (FFNN) to generate output probabilities of twenty audio events for each frame. Mel scale filter bank (FBANK) features are extracted from each frame, and its five consecutive frames are combined as one vector which is the input feature of the FFNN. The output layer of FFNN produces audio event probabilities for each input feature vector. More than five consecutive frames of which event probability exceeds threshold are detected as an audio event. An audio event continues until the event is detected within one second. The proposed method achieves as 71.8% accuracy for 20 classes of the UrbanSound8K and the BBC Sound FX dataset.