• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.6
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• pp.509-523
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• 2012

The sound quality based on design optimization, throughout the development process of various electronic office equipments, needs to be considered in order to respond the increased needs for the emotional satisfaction of customers in terms of psycho-acoustics. This paper focuses on how to describe the characteristics of operating sound radiated from laser printers by using various sound attributes, and to model the sound quality index that can properly evaluate the subjective preference on modification conditions in the improvement study quantitatively. Especially, the proposed verification process, in the form of combining the correlation based method and the decision error based method, was applied to improve the generality and reliability of a group of participants in the jury evaluation. The modified Aures tonality model was also proposed to improve the correlation coefficient with the mean response of participants by optimizing some parameters. As a result, the loudness, articulation index, roughness, tonality, fluctuation strength were used to model the sound quality index for laser printers by using the multiple-linear regression method. Through the improvement study, it was confirmed that replacing the absorbing materials is effective to reduce the tonalness radiated from the side of a reference printer model. Based on above results, it can be concluded that the proposed model has enough usefulness as quantitative evaluation index to evaluate the difference between modification conditions in the improvement study.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.258-264
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• 2010

In this paper, we propose an objective quality feature which is based on the human auditory system to measure vacuum cleaner noise. It is observed that some frequency bands are more sensitive to the human auditory system. Therefore, we divided the audible frequency range of vacuum cleaner noise into a number of frequency bands and the average energy of these bands was calculated. Among a number of average energies, an average energy of a frequency band was selected as the proposed feature. In order to test the performance of the proposed feature, fourteen vacuum cleaners were chosen and the noise was recorded in an anechoic-chamber. Then we performed subjective tests to obtain subjective scores of the noise data using the PCM (paired comparison method) and ACR (absolute category rating) subjective methods. The proposed objective quality feature shows high correlation with the subjective scores.

• 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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.2
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• pp.108-116
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• 2014

With the increasing the importance of emotional quality of vehicle, the sound quality of systems with electric motor components has become increasingly important. Electric motors are used for windows, seats, sun roof, mirrors, steering columns, windshield wiper and climate control blowers, etc. In this paper, a study was conducted to identify sound quality factors that contribute to customer's satisfaction and preference of the window lift system. Jury test for subjective evaluation was carried out and sound quality index was developed. Averaged sound pressure level and sharpness were significant factors when glass moves down. Also, maximum loudness at stop section and averaged loudness were significant factor when glass moves up. Next, noise source identification was carried out using beam forming method during glass transferred section and impulsive noise at stop section. Several improvement methods were applied using the source identification result. And finally, the degree of sound quality improvement was judged using sound quality index.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.232-237
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• 2013

The role of the window length of time-frequency transformation is important for the audio data hiding methods utilizing phase information. In this paper, the experiments for our audio data hiding method were conducted in order to evaluate the audio quality and robustness against reverberant environment. The experimental results showed the tendency that the worse audio quality but better robustness were obtained when the lengthy window was applied. The important reason for quality degradation was pre-echo which flatters the percussive sound. The results also indicated that the wireless communication theory related to the length of time-frequency transform can be applied in the field of audio data hiding and acoustic data transmission.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.309-309
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• 2010

한국은 산과 강이 많고 평야가 적은 지형적 특성상 터널 구간, 교량 구간 등이 많고 철도 차량이 이러한 구간들을 통과할 때 차량 내부에서 발생되는 과도 소음 또한 자주 발생하게 된다. 정상상태에서 소음에 비하여 과도상태에서 소음은 특성이 급격히 변화하여 승객들에게 더욱 큰 성가심을 유발할 수 있다. 특히 국내에서 운행중인 KTX 와 KTX 산천과 같은 고속철도의 경우 빠른 운행 속도로 인하여 과도 소음의 정도 또한 높아지게 된다. 따라서 더욱 정숙한 승객들의 승차 환경을 조성하기 위하여 과도상태에서의 실내소음 특성분석은 그 중요성이 더욱 커지고 있다. 본 연구에서는 고속철도가 감속 가속할 때, 터널을 통과할 때, 교량 위를 운행할 때, 차량이 교행할 때 등의 과도 상태에서의 실내 소음 특성을 음질평가의 방법으로 분석하고 정상상태에서의 소음 특성과 비교하였다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.831-834
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• 2000

본 논문은 주파수 전이신호와 시간 전이 신호에 대해서 고조파 잡음 여기 방법과 시간 분리 여기 방법을 적용한 2.4kbps 음성부호화 방법을 제안한다. 혼합 여기 부호화 방법은 주기 신호와 비 주기 신호를 효과적으로 표현하기 위해 하모닉 잡음 모델을 사용한다. 혼합신호에 대한 잡음 성분은 캡스트럴 분석 방법을 사용함으로써 추출되고, AR(Autoregressive Model) 모델에 의해 표현된다. 시간 전이구간 신호에서의 모호한 음성을 효과적으로 제거하기 위한 또 다른 방법이 제안된다. 제안된 시간 분리 방법은 시간 에너지 변화정도를 관찰함으로써 전이 시점을 감지하고 다른 시간 길이를 가지는 두 블록으로 분리하여 분석한다. 시간 분리 방법은 분석을 위한 비대칭 윈도우와 합성에서의 위상 합성 방법을 포함한다. 제안된 방법을 사용한 2.4kbps 음성부호화 방법은 주관적 음질 평가에서 전이구간에서의 지각적 음질의 향상을 보여주었으며, 원본 음성 스펙트럼과의 고조파 비 매칭에 의한 윙윙거리는 기계적인 잡음을 감소시킨다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.53-58
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• 2000

본 논문은 주파수 전이신호와 시간 전이 신호에 대해서 고조파 잡음 여기 방법과 시간 분리 여기 방법을 적용한 2.4 kbps 음성부호화 방법을 제안한다. 혼합 여기 부호화 방법은 주기 신호와 비 주기 신호를 효과적으로 표현하기 위해 하모닉 잡음 모델을 사용한다. 혼합신호에 대한 잡음 성분은 캡스트럴 분석 방법을 사용함으로써 추출되고, AR (Autoregressive Model) 모델에 의해 표현된다. 시간 전이구간 신호에서의 모호한 음성을 효과적으로 제거하기 위한 또 다른 방법이 제안된다. 제안된 시간 분리 방법은 시간 에너지 변화정도를 관찰함으로써 전이 시점을 감지하고 다른 시간 길이를 가지는 두 블록으로 분리하여 분석한다. 시간 분리 방법은 분석을 위한 비대칭 윈도우와 합성에서의 위상 합성 방법을 포함한다. 제안된 방법을 사용한 2.4 kbps 음성부호화 방법은 주관적 음질 평가에서 전이구간에서의 지각적 음질의 향상을 보여주었으며, 원본 음성 스펙트럼과의 고조파 비 매칭에 의한 윙윙거리는 기계적인 잡음을 감소시킨다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.129-132
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• 2009

Frame erasure is one of the most difficult problems in voice over IP (VoIP) networks and is a major source of speech quality degradation. In this paper, a frame erasure concealment algorithm based on nonlinear regression analysis is presented to minimize speech quality deterioration in code-excited linear prediction (CELP) based coders. We applied the proposed scheme to the ITU-T G.729 standard and obtained improved perceptual evaluation of speech quality (PESQ) scores compared to the conventional methods.