• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2003년도 10월 학술대회지
• /
• pp.185-188
• /
• 2003

In this paper, we construct phonetic GMM for text-independent speaker identification system. The basic idea is to combine of the advantages of baseline GMM and HMM. GMM is more proper for text-independent speaker identification system. In text-dependent system, HMM do work better. Phonetic GMM represents more sophistgate text-dependent speaker model based on text-independent speaker model. In speaker identification system, phonetic GMM using HMM-based speaker-independent phoneme recognition results in better performance than baseline GMM. In addition to the method, N-best recognition algorithm used to decrease the computation complexity and to be applicable to new speakers.

• 한국지능시스템학회논문지
• /
• 제17권5호
• /
• pp.696-700
• /
• 2007

본 논문에서는 화자 독립 음소 인식기를 사용하는 음소열 기반 음성 인식 시스템의 성능을 향상시키는 새로운 방법을 제안하였다. 화자독립 음소 HMM을 사용하는 음성 인식 시스템은 입력 문장에 대한 음소열만을 사용하므로 저장 공간은 크게 줄일 수 있다. 그러나 시스템의 성능은 화자독립 모델을 사용하므로 발생하는 음소 오차 때문에 화자 종속 시스템보다 저하된다. 여기에서는 화자 적응 기술을 사용하여 화자독립 모델과 학습 데이터간의 불일치를 감소시키도록 음소열과 변환 벡터를 반복적으로 추정하는 학습 방법을 제안하였다. 화자 적응을 위한 변환 벡터를 추정하기 위하여 확률적 매칭 방법이 사용되었다. 실험은 전화선을 통하여 얻어진 데이터를 사용한 실험에서 기존 방법에 비하여 약 45%정도 오차가 감소되었다.

• 대한음성학회지:말소리
• /
• 제52호
• /
• pp.111-120
• /
• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.

• 대한음성학회지:말소리
• /
• 제45호
• /
• pp.107-115
• /
• 2003

An efficient method that compensates the HMM parameters for the noisy speech recognition is proposed. Instead of assuming some analytical approximations as in the PMC, the proposed method directly re-estimates the HMM parameters by the segmental k-means algorithm. The proposed method has shown improved results compared with the conventional PMC method at reduced computational cost.

• 대한음성학회지:말소리
• /
• 제58호
• /
• pp.67-81
• /
• 2006

For a large corpus of time-aligned data, HMM based approaches are most widely used for automatic segmentation, providing a consistent and accurate phone labeling scheme. There are two methods for training in HMM. Flat starting method has a property that human interference is minimized but it has low accuracy. Bootstrap method has a high accuracy, but it has a defect that manual segmentation is required In this paper, a new algorithm is proposed to minimize manual work and to improve the performance of automatic segmentation. At first phase, voiced, unvoiced and silence classification is performed for each speech data frame. At second phase, the phoneme sequence is aligned dynamically to the voiced/unvoiced/silence sequence according to the acoustic phonetic rules. Finally, using these segmented speech data as a bootstrap, phoneme model parameters based on HMM are trained. For the performance test, hand labeled ETRI speech DB was used. The experiment results showed that our algorithm achieved 10% improvement of segmentation accuracy within 20 ms tolerable error range. Especially for the unvoiced consonants, it showed 30% improvement.

• 대한음성학회지:말소리
• /
• 제43호
• /
• pp.163-178
• /
• 2002

In this paper, we discuss the characteristics of Segmental-Feature HMM and summarize previous studies of SFHMM. There are several approaches to reduce the number of parameters in the previous studies. However, if the number of parameters decreased, the performance of systems also fell. Therefore, we consider the fast computation approach with preserving the same number of parameters. In this paper, we present the new segment comparison method to speed up the computation of SFHMM without loss of performance. The proposed method uses the three-frame calculation rather than the full(five) frames in the given segment. The experimental results show that the performance of the proposed system is better than that of the previous studies.

• 대한음성학회지:말소리
• /
• 제64호
• /
• pp.121-135
• /
• 2007

Speech signal is distorted by channel characteristics or additive noise and then the performances of speaker or speech recognition are severely degraded. To cope with the noise problem, we propose a modified HMM decoder algorithm using SNR-based observation confidence, which was successfully applied for GMM in speaker identification task. The modification is done by weighting observation probabilities with reliability values obtained from SNR. Also, we apply PSO (particle swarm optimization) method to the confidence function for maximizing the speaker identification performance. To evaluate our proposed method, we used the ETRI database for speaker recognition. The experimental results showed that the performance was definitely enhanced with the modified HMM decoder algorithm.

• 대한음성학회지:말소리
• /
• 제68권
• /
• pp.65-74
• /
• 2008

A statistical parametric speech synthesis system based on the hidden Markov models (HMMs) has grown in popularity over the last few years, because it needs less memory and low computation complexity and is suitable for the embedded system in comparison with a corpus-based unit concatenation text-to-speech (TTS) system. It also has the advantage that voice characteristics of the synthetic speech can be modified easily by transforming HMM parameters appropriately. In this paper, we present experimental results of voice characteristics conversion using the HMM-based Korean speech synthesis system. The results have shown that conversion of voice characteristics could be achieved using a few sentences uttered by a target speaker. Synthetic speech generated from adapted models with only ten sentences was very close to that from the speaker dependent models trained using 646 sentences.

• 제어로봇시스템학회논문지
• /
• 제16권4호
• /
• pp.335-338
• /
• 2010

A new method that improves the performance of large scale voice dialling system is presented using speaker adaptation. Since SI (Speaker Independent) based speech recognition system with phoneme HMM uses only the phoneme string of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the speaker dependent system due to the mismatch between the input utterance and the SI models. A new method that estimates the phonetic string and adaptation vectors iteratively is presented to reduce the mismatch between the training utterances and a set of SI models using speaker adaptation techniques. For speaker adaptation the stochastic matching methods are used to estimate the adaptation vectors. The experiments performed over actual telephone line shows that proposed method shows better performance as compared to the conventional method. with the SI phonetic recognizer.

• 정보처리학회논문지B
• /
• 제9B권2호
• /
• pp.215-222
• /
• 2002

본 논문에서는 음절이 잘 발달되어 있는 한국어에 대해서 신뢰할 수 있는 완전 자동화된 레이블링 시스템을 제안한다. 음운 및 음향학적인 정보를 최대한 이용하고 분할에러를 줄이기 위해서 조절 메카니즘의 하나로 DAC개념을 사용하여 음성을 speechlet으로 나누고 분할 된 음성 구간에 대해서 레이블링을 시도하는 DAC기반 분할알고리즘이다. HMM방법이 획일적이고 확정적인 성능을 갖는 반면 본 제안 방법은 음성학적인 특화지식을 컴포넌트로 개발 추가 계속 향상시킬 수 있는 프레임워크를 제시하고 있다는 점에서 주요 의의가 있다고 하겠다. MM과 같은 통계학적인 방법을 이용하지 않고 음운학적, 음향학적 지식만을 이용하는 새로운 방법은 수행속도와 음성학적인 특화 지식컴포넌트를 확장함에 따라 일관성이 있으며 효과적 방법으로 적용가능 할 것이다. 제안 방법을 검증하기 위하여 실험결과를 제시하였다.