• Journal of Digital Contents Society
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• v.8 no.3
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• pp.341-346
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• 2007

The purpose of this paper is to propose the design of VADSM(Value-Added Digital Speaker Module) which tunes up the speaker unit by measuring the speaker's frequency responses and controlling EQ band. This module can reduce audible distortions at particular frequency band and improve some flatness in the speaker's frequency response. VADSM is composed of DSP AMP and speaker unit. When a speaker transforms electrical signal to sound, the magnitude response at some frequencies are more or less than normal level. So, DSP AMP can be used to adjust those magnitudes up or down by controlling its EQ bands.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.74-80
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• 2010

Speech and audio codecs have been developed based on different type of coding technologies since they have different characteristics of signal and applications. In harmony with a convergence between broadcasting and telecommunication system, international organizations for standardization such as 3GPP and ISO/IEC MPEG have tried to compress and transmit multimedia signals using unified codecs. MPEG recently initiated an activity to standardize the USAC (Unified speech and audio coding). However, USAC RM (Reference model) software has been problematic since it has a complex hierarchy, many useless source codes and poor quality of the encoder. To solve these problems, this paper introduces a new RM software designed with an open source paradigm. It was presented at the MPEG meeting in April, 2010 and the source code was released in June.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.965-974
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• 2007

In this paper, we proposed the feature selection algorithm for multi-class genre classification. In our proposed algorithm, we developed GMM separation score based on Gaussian mixture model for measuring separability between two genres. Additionally, we improved feature subset selection algorithm based on sequential forward selection for multi-class genre classification. Instead of setting criterion as entire genre separability measures, we set criterion as worst genre separability measure for each sequential selection step. In order to assess the performance proposed algorithm, we extracted various features which represent characteristics such as timbre, rhythm, pitch and so on. Then, we investigate classification performance by GMM classifier and k-NN classifier for selected features using conventional algorithm and proposed algorithm. Proposed algorithm showed improved performance in classification accuracy up to 10 percent for classification experiments of low dimension feature vector especially.

• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.400-406
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• 2014

This paper describes a robust watermark decoding model using a SVM(Support Vector Machine). First, the embedding process is performed inversely for a watermarked signal. And then the watermark is extracted using the proposed model. For SVM training of the proposed model, data are generated that are watermarks extracted from sounds containing watermarks by four different embedding schemes. BER(Bit Error Rate) values of the data are utilized to determine a threshold value employed to create training set. To evaluate the robustness, 14 attacks selected in StirMark, SMDI and STEP2000 benchmarking are applied. Consequently, the proposed model outperformed previous method in PSNR(Peak Signal to Noise Ratio) and BER. It is noticeable that the proposed method achieves BER 1% below in the case of PSNR greater than 10 dB.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.191-199
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• 2006

This paper presents an efficient pre echo control scheme for MPEG Audio coders based on the psychoacoustic model II (PAM-II). Pre echo control is the final step for the calculation of masking threshold in the PAM II. It is to minimize the spread of quantization error over the processing frame. In the conventional encoders, pre echo is reduced by restricting the estimated masking threshold not to exceed the one obtained in the previous frame. The conventional method performs pre echo control not only for short blocks but also for long blocks, which lowers the masking threshold in long blocks and, in turn, increases the quantization noise level of corresponding blocks. This paper proposes an efficient pre echo control process. The test result shows a mean enhancement of more than 0.4 especially for complex signals on the ITU R 5 point audio impairment scale.

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.410-418
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• 2012

Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

• Journal of Internet Computing and Services
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• v.18 no.1
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• pp.57-64
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• 2017

Recent automobile manufacturing technology has improved not only the function and performance of cars, but also the audio systems in cars so as to increase their marketability. Automobile manufacturers always have the option of simply installing an expensive acoustic system to help customers enjoy a high-level sound quality car audio system. However, this also tends to increase the MSRP (Manufacturer's Suggested Retail Price) of the car. Therefore, it is desirable, where possible, to enhance the sound quality of plainer, less expensive audio devices to help customers feel as if they have a high-quality and expensive audio device in their car. In order to make this happen, the manufacturer must develop an optimal interior environment and audio system at a relatively lower cost. To this end, features of the car audio system can be enhanced by analyzing audio frequency response and using performance metrics to figure out the characteristics of the human auditory system. This study analyzed the sound field of Korean Recreation Vehicles (RVs) using the Internet of Things (IoT) sensor for the measurement of car audio system. As a result, high energy of sensitive bandwidth, one of the human auditory characteristics often makes annoying sound. This study also found that increasing the frequency response flatness is required by taking human auditory field into account when designing the car audio system for the future.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.270-277
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• 2010

In this paper, we proposed an algorithm that improves tone quality of noisy audio signals by using ICA(Independent Component Analysis) algorithm and perceptual filters. Many algorithms have been proposed to eliminate the noise from the audio signals, such as spectral subtraction method, perceptual filter, etc. The perceptual filter uses a noise that is acquired from silent ranges in the input signal. In this case, the improvement rate of tone quality decreases if the noise energy is changed by the environmental variation in a signal frame. But the proposed method estimates a noise that is changed at each frame using ICA algorithm. The estimated noise is applied to perceptual filter. To show the performance of the proposed algorithm, several tests are performed to various input signals. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR), noise-to-mask ratio (NMR) and Degradation Category Rating (DCR) test.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.119-122
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• 2000

인터넷이 일상생활에 다양하게 활용되면서 인터넷 채널을 통한 정보의 형태는 문자와 이미지 외에 음성, 오디오 신호 및 동영상 부분까지 확대되고 있다. 본 논문에서는 MPEG4-CELP를 인터넷 화상 통신의 음성 코덱용으로 사용하기 위한 최적화 기법 및 알고리듬의 개선을, DSP칩이 내장된 보드가 아닌 인터넷의 터미널로 사용되고 있는 펜티엄 프로세서를 장착한 PC에 초점을 맞추어 수행하였다. MPEG4-CELP VM C소스를 분석 및 프로파일(Profile)한 결과를 토대로 패라미터 추출을 위해 많은 연산을 수행하는 부호화기에 대해서 CPU상에 부하를 많이 주는 함수들을 제 1차 최적화 대상 함수들로 선정하고, CPU에 부하를 많이 주지는 않으나 호출되는 회수가 많은 함수를 2차 최적화 대상 함수로 선정해, C소스 레벨의 소프트웨어 파이프 라이닝(Software Pipelinging) 기법들을 적용하여 최적화를 수행하였다. 또한 1차 최적화 대상 함수의 경우에는 소프트웨어 파이프라이닝의 적용과 함께 연산량 감소를 위한 알고리듬 변형까지 수행하였다. 위의 과정을 거쳐 최적화 된 MPEG4-CELP는 펜티엄Ⅲ 450㎒ PC에서 음성을 부호화 하는데 원 VM소스에 비해 약 2배정도의 시간이 단축되는 것을 확인하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.11a
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• pp.24-27
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• 2013

본 연구는 지상파TV의 디지털전환 완료 이후 아날로그 AM과 FM라디오의 디지털 전환과 새로운 UHDTV방송 도입을 위해 VHF 상위대역과 700MHz 대역에서의 효율적인 주파수 할당과 활용방안을 제시하고 있다. 아울러 방송은 UHD 영상과 디지털오디오 등 고품질 서비스도 중요하지만 무엇보다 수신이 잘되고 편리해야 하므로 강인한 신호 전송과 수신환경 개선 방안도 함께 제시하고자 한다. 특히 VHF 상 하위 대역과 AM/FM 라디오 대역 및 DTV 대역 그리고 700MHz 대역 등 방송주파수 전체 대역에서 UHDTV방송과 디지털라디오방송 환경을 구축하는데 서로 충돌 없는 합리적인 주파수 할당과 SFN 전송망 구축에 의한 효과적인 주파수 이용 방안을 제시하고 있다.