• ETRI Journal
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• 제31권4호
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• pp.408-418
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• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• 말소리와 음성과학
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• 제15권4호
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• pp.45-51
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• 2023

나이가 들면서 발생하는 음성 문제는 사회적, 정서적으로 영향을 미칠 수 있으며, 나아가 고립감과 우울증으로 이어질 수 있다. 이에 본 연구에서는 노화로 인한 음향학적 특성 변화를 음질과 구어 유창성의 변화를 알아보고자 한다. 이를 위해 노년층 남성 20명과 청년층 남성 20명이 산출한 연장발성과 구절 읽기 과제를 녹음하여 분석하였다. 음질 분석 변수로 기본주파수(F0), 주기 변동률(jitter), 진폭 변동률(shimmer), 켑스트럼 정점(cepstral peak prominence, CPP) 값을 분석하였으며 구어 유창성 분석 변수로는 평균 음절 길이(average syllable duration, ASD), 조음 속도(articulation rate, AR), 구어 속도(SR)를 분석하였다. 연구결과, 음질 측정에서 노년층의 경우 F0가 높게 나타났으며 jitter, shimmer, CPP의 결과값을 통해 음질이 저하된 것으로 나타났다. 구어 유창성 분석 결과, 노년층은 ASD, AR, SR의 결과값을 통해 느리게 발화하는 것으로 나타났다. 음질과 구어유창성 간 상관관계 분석 결과, shimmer와 CPP 값과 각각 ASD와 SR에서 높은 상관관계가 나타났다. 본 연구결과를 통해 노화에 따른 음성과 구어 유창성 변화를 조기에 발견하고 이에 대한 적절한 훈련법을 제공할 수 있을 것으로 기대된다.

• Journal of information and communication convergence engineering
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• 제5권2호
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• pp.177-180
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• 2007

We compared CS-ACELP with QCELP speech coder in CDMA cellular under channel error environment and experimented performance with its measured value under channel error environment. Also, we specified the effective coding scheme to overcome. CS-ACELP speech coder using a LSP vector quantizer shows transparent speech quality from the results that SD is 0.92dB and outlier frames over 2dB is 2.9% in the BER 0.10% condition. CS-ACELP speech coder which is utilizing MA predictor shows better results on SVR and SEGSNR than QCELP speech coder(IS-96) adopting DPCM type predictor when bit error occurs from BER 0.01% to 0.50%.

• ETRI Journal
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• 제38권2호
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• pp.366-375
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• 2016

This paper describes an algorithm to suppress composite noise in a two-microphone speech enhancement system for robust hands-free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal-dominant time-frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech-dominant TFBs are identified among the previously detected nonstationary signal-dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin-wise output signal-to-noise ratio is obtained with these power estimates and a Wiener post-filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post-filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.

• 음성과학
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• 제14권2호
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• pp.91-103
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• 2007

The purpose of this paper is to develop a model enabling to predict the prosody of Korean text-to-speech synthesis using the CART and SKES algorithms. CART prefers a prediction variable in many instances. Therefore, a partition method by F-Test was applied to CART which had reduced the number of instances by grouping phonemes. Furthermore, the quality of the text-to-speech synthesis was evaluated after applying the SKES algorithm to the same data size. For the evaluation, MOS tests were performed on 30 men and women in their twenties. Results showed that the synthesized speech was improved in a more clear and natural manner by applying the SKES algorithm.

• 말소리와 음성과학
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• 제5권4호
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• pp.33-41
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• 2013

This study investigated the perceptual and cepstral/spectral characteristics of phonation and their relationships in dysarthria in connected speech. Twenty-two participants were divided into two groups; the eleven dysarthric speakers were paired with matching age and gender healthy control participants. A perceptual evaluation was performed by three speech pathologists using the GRBAS scale to measure the cepstrual/spectral characteristics of phonation between the two groups' connected speech. Correlations showed dysarthric speakers scored significantly worse (with a higher rating) with severities in G (overall dysphonia grade), B (breathiness), and S (strain), while the smoothed prominence of the cepstral peak (CPPs) was significantly lower. The CPPs were significantly correlated with the perceptual ratings, including G, B, and S. The utility of CPPs is supported by its high relationship with perceptually rated dysphonia severity in dysarthric speakers. The receiver operating characteristic (ROC) analysis showed that the threshold of 5.08 dB for the CPPs achieved a good classification for dysarthria, with 63.6% sensitivity and the perfect specificity (100%). Those results indicate the CPPs reliably distinguished between healthy controls and dysarthric speakers. However, the CPP frequency (CPP F0) and low-high spectral ratio (L/H ratio) were not significantly different between the two groups.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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• pp.307-310
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• 2007

In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed to improve the quality of decoded speech under a burst packet loss condition. The proposed algorithm is based on the recovery of voiced excitation using an estimate of the voicing probability and the generation of random excitation by permutating the previously decoded excitation. The voicing probability is estimated from the correlation using the previous correctly decoded excitation and pitch. The proposed algorithm is implemented as a PLC algorithm for G.729 and its performance is compared with PLC employed in G.729 by means of perceptual evaluation of speech quality (PESQ) and an A-B preference test under the random and burst packet losses with rates of 3% and 5%. It is shown that the proposed algorithm provides better speech quality than the PLC of G.729, especially under burst pack losses.

• 한국인터넷방송통신학회논문지
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• 제9권5호
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• pp.129-132
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• 2009

프레임 손실은 VoIP 망에서의 음질 저하의 주요 원인이다. 본 논문에서는 VoIP 망에서 주로 사용되는 CELP 기반 음성부호화기의 음질 저하를 최소화하기 위해 비선형 회귀분석 기반의 프레임손실은닉 알고리즘을 제안한다. 제안된 기법은 ITU-T G.729 표준 코덱에 적용되었으며, 기존 방법들에 비해 향상된 PESQ 성능을 보였다.

• 말소리와 음성과학
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• 제1권4호
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• pp.165-170
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• 2009

This study was to investigate for comparison between single rating and average ratings from multiple presentations of the same stimulus for measuring the voice quality of dysphonia using 7-point equal-appearing interval (EAI) rating scale. Overall severity of voice quality for 46 /a/ vowel stimuli (23 stimuli from dysphonia, 23 stimuli from control) was rated by 3 experienced speech-language pathologists (averaged 19 years; range = 7 to 40 years). For average ratings, each stimulus was rated five times in random order and averaged from two to five times. Although higher inter-rater reliability was found in average ratings than in single rating, there were no significant differences in rating scores between single and multiple average ratings judged by experienced listeners, suggesting that auditory perceptual ratings judged by well-trained listeners have relatively good agreement with the same stimulus across the judgment. Larger variations in perceptual ratings were observed for moderate voices than for mild or severe voices, even in the average ratings.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2232-2235
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• 2003

In speech communication systems where two different speech codecs are interoperated, transcoding algorithm is a good approach because of its low complexity and improved synthesized speech quality. This paper proposes an efficient method to further improve the performance of transcoding algorithms as well as to reduce the complexity. In the conventional transcoding algorithms. a post-filter and a perceptual weighting filter should be operated sequentially because both decoding and encoding processes are needed. This results in the redundancy of the processing in terms of complexity and perceptual quality. Using the fact that their filter structures are similar, we replaced the two filters with one. The proposed algorithm requires 72.8% lower complexity than the conventional transcoding algorithm when we compare only the complexity of the filtering processes. The results of both objective and subjective tests verify that the proposed algorithm has slightly better quality than the conventional one.