• 말소리와 음성과학
• /
• 제1권4호
• /
• pp.159-163
• /
• 2009

The purpose of this study is to conduct a correlational analysis among pitch, nasalance, and acoustic quality parameters estimated by two speech analysis softwares NasalView(version 1.31), Dr. Speech 4.5(Tiger Electronics). Thirty females and 25 males with normal voice participated in the study. The Pearson correlation coefficient was determined through a statistical analysis. The results came out as follows; Firstly, there was a correlation between $F_0$ and voice quality parameters, however there was no correlation between $F_0$ and nasalance. Secondly, nasalance showed a correlation with voice quality parameters.

• 한국음향학회지
• /
• 제18권6호
• /
• pp.38-43
• /
• 1999

SK Telecom에서 현재 서비스중인 EVRC 보코더는 유선전화 수준의 음질을 제공하는 우수한 음성 부호화기이나, 약전계에서 급격한 음질 저하를 보인다. 본 논문에서는 실제 서비스 상황에서 발생하는 EVRC 보코더의 음질 저하 현상 및 그 원인을 분석하였고, 해결책으로 피치 예측과 점진적 복원 기법을 제안하였다. 다양한 전파환경에 대한 음질 평가방법으로 선호도 실험을 수행하였고, 제안한 방법이 효과적임을 확인하였다.

• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2002년도 11월 학술대회지
• /
• pp.33-36
• /
• 2002

As the telematics service that is the integration of information technology approaches commercialization, the necessity and gravity of speech technology is rapidly growing. The speech technology occupies important position in the telematics service because it informs the starting of service and the retrieved result. This service must provide high accuracy of speech recognition and natural synthesis of human speech in a driving environment and it is especially true for the fee-for-service. For high quality TTS, the speech synthesis technique that makes optimal synthesis database and uses efficiently this database is required. In this paper, we describe the design of phonetically balanced sentences used for speech database, the selection of service-suitable-speaker, the extraction methods of accurate phoneme boundary, and the factors which are taken into consideration in the extraction stage of prosody. Finally we show the real case that has commercially implemented.

• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2004년도 춘계 학술대회 발표논문집
• /
• pp.235-238
• /
• 2004

Frame erasures cause speech quality degradation in wireless communication networks or packet networks. The degradation becomes worse when consecutive frame erasures occur. Speech coders have a frame erasure concealment(FEC) mechanism to compensate for frame erasures. It is meaningful to evaluate the performance of FEC mechanisms for frame erasures that occur in communications networks. In this paper, various frame erasures are designed. And the FEC algorithms of speech coders are evaluated and analyzed with the Perceptual Evaluation of Speech Quality(PESQ). It is found that the performances vary in accordance with frame erasure types, frame erasure rates, and utterance lengths.

• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2006년도 추계학술대회 발표논문집
• /
• pp.131-134
• /
• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.

• 음성과학
• /
• 제11권4호
• /
• pp.67-73
• /
• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2005년도 추계 학술대회 발표논문집
• /
• pp.63-66
• /
• 2005

For the Large Corpus based TTS the consistency of the speech corpus is very important. It is because the inconsistency of the speech quality in the corpus may result in a distortion at the concatenation point. And because of this inconsistency, large corpus must be tuned repeatedly One of the reasons for the inconsistency of the speech corpus is the different glottal characteristics of the speech sentence in the corpus. In this paper, we adjusted the glottal characteristics of the speech in the corpus to prevent this distortion. And the experimental results are showed.

• 한국지능시스템학회논문지
• /
• 제17권7호
• /
• pp.945-951
• /
• 2007

잡음의 제거는 음성을 인터페이스로 하는 기기들에 필수적이라고 할 수 있다. 실질적으로 통화 품질이나 음성 인식률은 음성 입력부의 주변에서 들어오는 원치 않는 가산성 잡음에 의해서 크게 열화된다. 본 논문에서는 기본적으로 두 개의 마이크로폰을 이용한 잡음제거 방법을 제안한다. 마이크를 여러 개 사용했을 때의 장점은 방향 정보를 이용할 수 있다는 것인데 이는 사람 목소리, 음악 소리 등의 비정상성 잡음을 제거하는 데에 유용하다. 제안된 잡음제거 알고리즘은 위너필터에 기반 한다고 볼 수 있다. 위너필터에 의한 잡음제거를 위해서는 검출하고자 하는 음성과 제거하고자 하는 잡음의 주파수 응답이 동시에 추정 가능해야 한다. 이를 위해서 주파수 영역에서 스펙트럼 분류를 시행하여 위너필터 기반의 잡음제거에 필요한 정보를 얻는다. 제안된 알고리즘을 이용한 성능은 잘 알려진 프로스트 (Frost) 알고리즘 및 적응 모드 컨트롤러를 갖는 generalized sidelobe canceller (GSC)와 비교하였다. 성능의 지표로는 객관적 음질 평가의 방법 중에서 널리 쓰이고 있는 perceptual evaluation of speech quality (PESQ) 및 음성 인식률이 사용되었다.

• 전자공학회논문지B
• /
• 제33B권2호
• /
• pp.88-94
• /
• 1996

In this paper, we present a half rate speech coder using trellis excitation. The coder combines code-excited linear prediction (CELP) system and trellis quantization method using the codebook expansion, and it produces higher speech quality than the typical CELP coder for the same transmission rate. A subjective comparison with 3~8 bit .$\mu$-law PCM indicates that the half rate coder provides speech quality between 5-bit and 6-bit $\mu$-law PCM .

• 대한음성학회지:말소리
• /
• 제55권
• /
• pp.143-158
• /
• 2005

The adequate compression algorithm is essential to achieve high quality embedded TTS system. in this paper, we Propose waveform interpolation coder for TTS corpus compression after many speech coder investigation. Unlike speech coders in communication system, compression rate and anality are more important factors in TTS DB compression than other performance criteria. Thus we select waveform interpolation algorithm because it provides good speech quality under high compression rate at the cost of complexity. The implemented coder has bit rate 6kbps with quality degradation 0.47. The performance indicates that the waveform interpolation is adequate for TTS DB compression with some further study.