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

• Journal of Broadcast Engineering
• /
• v.11 no.2 s.31
• /
• pp.200-206
• /
• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.48 no.5
• /
• pp.129-136
• /
• 2011

This paper presents the coding method that have multi-mode and efficiency of audio codecs using the feature of audio signal. Recently, the developed extension super-wideband codec based on G.718 wideband divides two mode between Generic and Sinusiodal. So codec efficently encode audio signal exist in super-wideband. But the codec is not as efficent coding for harmonic component of wind instrument and string instrument and individual-Line component of percussion instrument. The proposed method are modeling and encoding multiple pitch and individual-line feature using multi mode coding. For the performance evaluation, we used SNR in MDCT domain for objective test and MUSHRA test for subjective test. As a result, the performance of SNR and MUSHRA test of the proposed method have better performance than the G.718 super-wideband codec.

• Korean Journal of Digital Imaging in Medicine
• /
• v.10 no.2
• /
• pp.17-22
• /
• 2008

It is to improve customer satisfaction measurement and CT Scan process without delay of examination time when is using Scan positioning time(Planning time) that time is happened always between research reactor CT examination to increase fear and examination satisfaction by the customer's comprehension tribe which get the latest contrast enhancement CT examination. Needs and interests that customer wants to compose visual and auditory Contents to be played to Scan positioning time did questionnaire about curiosity later before CT examination to 600 people for October - November 2 months of 2006 to customer whole that get CT examination on source. Data getting through questionnaire investigated examination comprehension and satisfaction through questionnaire after experiment Scan Positioning to 500 coming to help customers to be source CT examination for 3 months February December - 2007 year in 2006 manufacturing Voice and Visual announce media for reference. To customer who interest degree appeared, and answers preparatory audit from preparatory audit about curiosity of CT examination customer to order of examination time required(43%), contrast media side effect(26%), examination region(20%), breath(10%), etc..(1.5%) audio-visual materials in questionnaire that attain after do reclamation among examination age, sex, reception type of irrelatively in 91% of target increase of hailing degree and examination satisfaction appear. Searched that customer hailing and satisfaction are increased greatly when use of audio-visual materials in satisfaction result that use CT Positioning delay time. In experiment process, It took lacking part by method that use hearing in case of do not use sight as is unavoidable in subject position or old age. Through this, audio-visual materials could analogize that it is more useful method that use sight and hearing at the same time.

• The Journal of the Acoustical Society of Korea
• /
• v.27 no.1
• /
• pp.18-25
• /
• 2008

In this paper, new lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder uses Huffman coding utilizing the statistical characteristics of spectral coefficients, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, new lossless coding scheme with better performance is proposed that consists of bit-plane transform and run-length coding. In the proposed scheme, the spectral coefficients are first transformed by bit-plane into 1-D bit-stream with better correlative properties, which is then coded intorun-length and is finally Huffman coded. In addition, the coding performance is further increased by applying the proposed bit-plane coding selectively to each group, after the entire frequency is divided into 3 groups. The performance of proposed coding scheme is measured in terms of theoretical number of bits based on the entropy, and shows at most 6% enhancement compared to that of conventional lossless coder used in AAC audio codec.

• Proceedings of the IEEK Conference
• /
• 2001.09a
• /
• pp.227-230
• /
• 2001

반향(Echo)을 이용한 워터마킹은 오디오 신호에 인위적인 반향을 첨가함으로써 정보를 삽입한다. 다른 오디오 워터마킹 방법과 마찬가지로 반향 오디오 워터마킹은 시간축 공격에 대해 강인하지 못한 단점을 가지고 있다. 특히, 오디오 신호의 피치를 보존하면서 재생 시간을 변형시키는 시간 스케일 변형 (Time Scale Modification)에 대해서는 별도의 방어를 위한 알고리듬이 없을 경우 전혀 복호화가 이뤄지지 않는다. 본 논문에서는 반향 오디오 워터마킹의 성능 향상을 위해 시간 스케일 변형 공격에 대응하여, 변형된 정도를 검출하고 보상하여 복호화가 가능하도록 하는 새로운 알고리듬을 제안한다.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.7-10
• /
• 2000

This paper focuses on the development of speech enhancement techniques for hands-free audio terminals, including two major problems : noise cancellation and acoustic echo cancellation. The objective is to find a joint structure to get a near-end speech signal with minimum distortion and low levels of echo and noise. To solve the two problems, a new promising technique is studied and tested in computer simulation conditions.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2021.06a
• /
• pp.65-66
• /
• 2021

본 논문에서는 Wave-U-Net 기반의 오디오 부호화 성능 향상 기법을 제안한다. 기존의 인공지능 기반 오디오 부호화 기술은 오디오의 주파수 정보를 복원하는 방식이기 때문에 완전한 복원을 위해서 주파수의 위상 정보를 별도로 부호화하여 전송해야 한다는 문제점이 있다. 따라서 본 논문에서는 오디오 부호화의 성능 향상을 위해 음원의 주파수 분석을 필요로 하지 않은 end-to-end 모델인 Wave-U-Net을 사용할 것을 제안한다. Wave-U-Net을 사용한 음원이 사용 전의 음원보다 객관적, 주관적 평가 지표에서 우수한 성능을 보이는 것을 확인하였다.

• The Journal of the Acoustical Society of Korea
• /
• v.27 no.1E
• /
• pp.17-24
• /
• 2008

Blind speech separation (BSS) is well-known as a powerful technique for speech enhancement in many real world environments. In this paper, we propose a new application of BSS - acoustic echo cancellation (AEC) in a car environment. For this purpose, we develop a block-online BSS algorithm which provides robust separation than a batch version in changing environments with moving speakers. Simulation results using real world recordings show that the block-online BSS algorithm is very robust to speaker movement. When combined with AEC, simulation results using real audio recording in a car confirm the expectation that BSS improves double talk detection and echo suppression.

• Journal of Broadcast Engineering
• /
• v.9 no.1
• /
• pp.9-16
• /
• 2004

In this paper, we present a speech enhancement method as a pre-processor for voice commander under car environment. For the friendly and safe use of voice commander in a running car, non-stationary audio signals such as music and non-candidate speech should be reduced. Ow technique is a two microphone-based one. It consists of two parts Blind Source Separation (BSS) and Kalman filtering. Firstly, BSS is operated as a spatial filter to deal with non-stationary signals and then car noise is reduced by kalman filtering as a temporal filter. Algorithm Performance is tested for speech recognition. And the results show that our two microphone-based technique can be a good candidate to a voice commander.