• Speech Sciences
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• v.6
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• pp.163-179
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• 1999

For the set up, management and repair of a mobile communication system, continuous estimation of speech quality is required. Speech quality measurement can be conducted by listener's judgement in a subjective test such as MOS (Mean Opinion Score) test. However, this method is laborious, expensive and time-consuming, it is advisable to predict subjective speech quality via objective measures. This paper presents a robust objective speech quality measure, PLP-CMS (Perceptual Linear Predictive-Cepstral Mean Subtraction), which can predict subjective speech quality in mobile communication systems. PLP-CMS has a high correlation with subjective quality owing to PLP (Perceptual Linear Predictive) analysis and shows a robust performance not being influenced by PSTN (Public Switched Telephone Network) channel effects due to CMS (Cepstral Mean Subtraction). To prove the performance of our proposed algorithm, we carried out subjective and objective quality estimation on speech samples which are variously distorted in a real mobile communication system. As a result, we demonstrated that PLP-CMS has a higher correlation with subjective quality than PSQM (Perceptual Speech Quality Measure) and PLP-CD (Perceptual Linear Predictive-Cepstral Distance).

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.02a
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• pp.125-128
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• 2008

IPTV의 상용화에 맞추어 사용자와 TV간의 정보 교환에 의한 대화형 서비스들이 제공되고 있으며, 특히 음성인식 기술은 이러한 서비스를 실현하기 위한 중요한 기술 중의 하나로 대두되고 있다. TV에서의 음성인식 수행을 위해서는 가정환경과 같은 제한된 공간에서 효율적으로 사용자의 음성을 TV에 전송할 수 있는 근거리 무선통신 수단이 필요하게 된다. 특히, 리모트 컨트롤러와 같은 저전력 시스템 환경에서 구현이 가능해야 한다. 따라서 이러한 제한된 조건에서 최적의 성능을 갖는 음성 전송 시스템 개발이 요구되고 있다. 본 논문에서는 블루투스 환경 하에서 음성인식을 위해 필요한 음성전송 시스템을 실시간 구현한다. 효율적인 음성전송을 위해 G.711을 기본 코덱으로 사용하며, 음성전송 시 발생하는 패킷손실에 따른 음성 품질 저하를 줄이기 위해 G.711 패킷손실 은닉 알고리즘을 음성전송 시스템에 적용한다. 특히 G.711 패킷 손실 은닉 알고리즘 수행을 위해 블루투스 프로토콜 스택application layer에 RTP 프로토콜을 적용하여 패킷 손실 여부를 확인하고, 패킷 손실 발생 시 패킷손실 은닉 알고리즘을 통해 음성의 품질 저하를 줄인다. 구현된 시스템의 성능을 평가한 결과, G.711 패킷 손실 알고리즘을 적용하여 2~10%의 패킷손실 환경에서 14.7%의 음질개선을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.110-114
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• 1997

When applying the nonuniform sampling to noisy speech signal, the required data rate increases to be comparable to or more than that by uniform sampling such as PCM. To solve this problem, we have proposed the waveform coding method, multiband nonuniform waveform coding(MNWC), applying the nonuniform sampling to band-separated speech signal[7]. However, the speech quality is deteriorated when it is compared to the uniform sampling method, since the high band is simply modeled as a Gaussian noise with average level. In this paper, as a good method to overcome this drawback, the high band is modeled as one of 16 codewords having different center frequencies. By doing this, with maintaining high speech quality as MOS score of average 3.16, the proposed method achieves 1.5 times higher compression ratio than that of the conventional nonuniform sampling method(CNSM).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.61-70
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• 2003

In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.21-24
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• 2001

본 논문은 신경 회로망을 이용하여, 전화망 대역의 음성, 즉, narrowband 음성에서 wideband 음성을 복원하고자 했다. BP 알고리즘을 사용하는 기존의 신경 회로망의 경우에는 음성과 같이 복잡하고 크기가 큰 훈련데이터에 대해서는 훈련이 제대로 되지 않는 단점이 있다. 그러므로 븐 논문에서는 이를 해결하기 위해 입력으로 들어온 LPC 켑스트럼 벡터를 k-means 알고리즘을 이용하여 미리 정한 개수의 cluster로 나눈 다음, 각각의 cluster에 대해 독립적인 신경 회로망을 적용했다 이로 인해 각각의 신경 회로망은 제한되고 서로 상관관계가 많은 음성들만 훈련하면 되므로, 기존의 신경 회로망에서 생기는 훈련의 정체를 개선할 수 있었다. 또 clustering 과정에서 생기는 오류를 보완하기 위해 후보신경 로망들의 출력에 fuzzy 개념을 적용해서 최종 출력을 내도록 했다 실험 결과에서, 제안한 알고리즘은 기존의 codebook mapping 알고리즘보다 스펙트럼 거리척도에 의한 비교 및 주관적인 음질 평가 양쪽에서 개선된 성능을 보였다.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.23 no.1
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• pp.73-83
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• 2013

In previous works, psychoacoustic parameters have been used for objective quantification. However, these parameters do not agree well with subjective assessment. Therefore, the correlation between psychoacoustic parameters and the subjective rating of door closing sounds of sampled cars is low, and it is not sufficient to use psychoacoustic parameters as an objective metric to quantify the sound quality of door closing sounds. In this paper, a new method is proposed to objectively quantify the sound quality based on physiological acoustics and statistical signal processing. The gammatone filter, as a pre-processing, is used in models of the auditory system and kurtosis, which is the fourth-order moment of temporal signal, and is used to extract information about sound quality quantification for door closing sounds. The new metric obtained through the proposed method is highly correlated with subjective rating, and it is successfully applied to the quantification of the sound quality of door closing sounds.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.4 s.109
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• pp.428-436
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• 2006

The assessment of an urban site depends on the way whether it responds to multiple needs such as functionality, aesthetic and complex comfort of acoustic, thermal, lighting and air quality etc. This study aims to investigate the assessment of various urban soundscapes in the sense of acoustic amenity by the questionnaires. As a result, acoustic amenity assessment was influenced by the non-acoustic factors such as environment assessment of visual, thermal, air quality etc. In the sense of sound quality, urban environmental sound was interpreted as 3 factors of strength, evolution of time, spacial localization. So these factors would be considered in the new assessment method for acoustic amenity. And it was shown that the subjects tend to perceive the noise level less than $3{\sim}5dB\;L_{eqA}$ according to the urban landscapes under the similar noise exposure level.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.11
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• pp.1137-1143
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• 2012

This paper presents the correlation between psychological and physiological acoustics for the automotive sound. The research purpose of this paper is to evaluate the sound quality of turn-signal sound of a passenger car based EEG signal. The previous method for the objective evaluation of sound quality is to use sound metrics based on psychological acoustics. This method uses not only psychological acoustics but also physiological acoustics. For this work, the sounds of 7 premium passenger cars are recorded and evaluated subjectively by 30 persons. The correlation between this subjective rating and sound metrics is calculated based on psychological acoustics. Finally the correlation between the subjective rating and the EEG signal measured on the brain is also calculated. Throughout these results the new evaluation system for the sound quality on interior sound of a passenger car has been developed based on bio-signal.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.327-330
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• 2002

디지털 워터 마킹은 일반적으로 지적 재산권의 보호를 위해서 컨텐츠 소유자가 자신의 컨텐츠에 저작권 정보를 나타내는 워터마크를 삽입하여 제 3자가 허가 없이 디지털 컨텐츠의 사용을 할 경우 불법적인 사용을 막아내는 데 사용된다. 본 논문에서는 워터마킹 기법의 새로운 적용 예로써 인터넷 또는 무선 환경을 통한 멀티미디어 데이터 전송 시 수신된 데이터의 품질을 사용자 측면에서 측정 할 수 있는 방안, 즉 디지털 워터마킹 기법의 새로운 적용 예에 대하여 고찰하였다. 제안하는 방법은 네트워크 상에서 발생하는 패킷 손실, 지연 등의 통계적 해석으로 수신 된 멀티미디어 데이터의 품질을 평가하는 척도로 사용하는 기존의 방법과는 달리 사용자 측면에서 눈으로 보이는 데이터의 일그러짐 정도나, 음질의 왜곡을 정량적인 측정치로 보여 줄 수 있음으로써, 오디오 데이터의 QoS(Quality of Service)를 보장하기 위한 수단이나 수신품질에 따른 차별화 된 과금 결정 등에 활용할 수 있을 것이다. 이를 위해 본 논문에서는 멀티미디어 데이터의 품질 측정을 위한 블라인드 워터마킹 기법을 적용하고 다양한 비트 에러율 (Bit Error Rate, BER)을 갖는 BSC(Binary Symmetric Channel)에 적용시켜 봄으로써 멀티미디어 데이터에 삽입된 워터마크 신호의 검출정도와 수신된 미디어 신호의 PSNR비가 선형적인 분포를 가지게 됨을 확인 할 수 있었다.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.7 no.5
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• pp.90-96
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• 2008

This paper presents an algorithm of a single chip acoustic speech enhancement for telematics device. The algorithm consists of two stages, i.e. noise reduction and echo cancellation. An adaptive filter based on cross spectral estimation is used to cancel echo. The external background noise is eliminated and the clear speech is estimated by using MMSE log-spectral magnitude estimation. To be suitable for use in consumer electronics, we also design a low cost, high speed and flexible hardware architecture. The performance of the proposed speech enhancement algorithms were measured both by the signal-to-noise ratio(SNR) and recognition accuracy of an automatic speech recognition(ASR) and yields better results compared with the conventional methods.