• Proceedings of the IEEK Conference
• /
• 2003.11c
• /
• pp.172-175
• /
• 2003

In this paper we propose a method fur the evaluation of the quality of service for VoIP services in NGN. Specifically, let us anatomize the elements of delay of a voice connection in the network in an end-to-end manner and investigate expected value at each point. We extract the delay time in each element in the network such as gateway, network node, and terminal equipment, and estimate an upper bound fur the tolerable delay in each element.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06c
• /
• pp.171-174
• /
• 1998

본 논문에서는 음성합성(speech synthesis) 및 부호화(coding) 시스템에 있어서 음원(voice source) 모델링에 관한 문제를 살펴보고자 한다. 기존의 음원 모델링 시스템이 가지고 있는 여러 문제들을 극복하고자 기저함수(basis function) 의 가중 합(weighted-sum)으로 음원을 모델링 하는 새로운 기법을 제안하고자 한다. 제안한 방법에서는 음원 파형(voice source waveform)을 적절히 표현하기 위해서 필터뱅크(filter bank)에 기초한 기저함수의 가중 합으로 나타낸다. 다양한 음원 특성을 효과적으로 나타내는 음원 파라미터를 구하기 위하여 EM(estimate maximize)에 기초한 구조에 관해 조사한다. 제안한 방법을 이용하여 다양한 유성음에 대해 실험을 수행하였다. 실험결과 제안한 추정(estimation) 방법 및 모델링 방법을 이용하면 기존의 방법에 비해 더 정확한 음원 파형을 추정할 수 있고, 다양한 음원 특성을 나타낼 수 있다. 또한 음성합성 및 부호화에서도 음성품질(voice quality)를 개선시킬 수 있으리라 기대된다.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
• /
• v.19 no.2
• /
• pp.85-88
• /
• 2008

Several parameters are used for the assessment of phonatory function and voice quality in clinical settings. Glottic airflow, subglottal pressure, mean phonation time, laryngeal resistance and voice efficiency are the most commonly used aerodynamic parameters. Aerodynamic analysis is developed to evaluate phonatory energy source and to estimate laryngeal efficiency. Also these measurements have shown the good correlation with perceptions of breathiness and findings of glottic competence. Aerodynamic study is important to understand relationships between pulmonary and phonatory function.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.7
• /
• pp.10-18
• /
• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
• /
• v.22 no.2
• /
• pp.143-150
• /
• 2011

Background and Objectives : Clinically, as a tool for voice assessment before and after the operation or the voice treatment, acoustic analysis is widely used. However, in clinical situations, acoustic parameters vary according to how the assessment is made. Thus, with voice disease patients as subjects, we are to investigate what influence intensity increase exerts on acoustic parameters and how to reduce variation according to the way of assessing. Material and Method : At the voice clinic of the department of otorhinolaryngology in Gangnam Severance Hospital, with 30 female voice-disease patients (40.6 years old on the average) and 23 male voice-disease patients (40.1 years old on the average) as subjects, using the Dr Speech vocal-assessment program, we statistically tested the significance of the difference in each of acoustic parameters between when the "Ah" vowel is produced with a normal voice and when the "Ah" vowel is produced with a loud voice. Results : Acoustic parameters that showed a statistically significant difference according to intensity increase were Jitter, SD F0, and NNE for females, and Jitter, SD F0, HNR, SNR, and NNE for males. Voice quality estimates showed a statistically significant difference according to intensity increase in female hoarse voice, female breathy voice, and male breathy voice. Conclusion : In this research, acoustic analysis, which is generally used for voice assessment before and after the operation or the voice treatment, showed a tendency that acoustic parameters became better under the influence of intensity increase except for the cases where a voice disease was severe. Thus, to raise the reliability of voice assessment, the range of intensity needs to be set up. This should be the topic for the future research.

• Proceedings of the KIEE Conference
• /
• 2007.07a
• /
• pp.1797-1798
• /
• 2007

In field of voice pathology, diverse statistics extracted form pitch estimation were commonly used to assess voice quality. In this study, we proposed robust pitch detection algorithm which can estimate pitch of pathological voices in benign vocal fold lesions. we also compared our proposed algorithm with three established pitch detection algorithms; autocorrelation, simplified inverse filtering technique, and nonlinear state-space embedding methods. In the database of total pathological voices of 99 and normal voices of 30, an analysis of errors related with pitch detection was evaluated between pathological and normal voices, or among the types of pathological voices. According to the results of pitch errors, gross pitch error showed some increases in cases of pathological voices; especially excessive increase in PDA based on nonlinear time-series. In an analysis of types of pathological voices classified by aperiodicity and the degree of chaos, the more voice has aperiodic and chaotic, the more growth of pitch errors increased. Consequently, it is required to survey the severity of tested voice in order to obtain accurate pitch estimates.

• The Journal of the Acoustical Society of Korea
• /
• v.32 no.6
• /
• pp.541-547
• /
• 2013

The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.

• The Journal of the Acoustical Society of Korea
• /
• v.29 no.2
• /
• pp.148-153
• /
• 2010

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. A Speech signal consists of Voice and Unvoiced Band in The MBE excitation model. And the energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced. We use the IMBE vocoder for the Voice or Unvoice band of segmented speech signal. Continuously we calculate the segmented SNR using that information and the energy of each band. And we estimate the SNR of continuous speech signal.

• Journal of Communications and Networks
• /
• v.9 no.4
• /
• pp.446-456
• /
• 2007

Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.39B no.8
• /
• pp.518-527
• /
• 2014

This study is to estimate the learning curve based on the consequences of reduced voice of customer from each telecommunication service products. We used Exponential Decay Model, which is the most popular among the learning curve models. We attempted to add how VOC changes in accordance with seasonal factors, human resource input, application of software, and the investment. The results of the empirical analysis of each service product as follows: First, as learning curve, customer complaints decreased. Second, human resource input, Network fault make increase or decrease customer complaints(VOC). Third, even though increasing the customer's quality of experience, VOC would not decrease due to service paradox.