• 전기전자학회논문지
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• 제27권4호
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• pp.472-476
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• 2023

피치 혹은 기본 주파수는 음성 신호의 주요 특성 인자이며 음성 부호화, 음성인식, 화자인식 등의 다양한 음성 관련 응용에 활용된다. 본 논문에서는 기본 주파수의 역수인 음성의 피치 주기를 추정하기 위해서 음성 신호의 변곡점을 이용한다. 변곡점은 국소적인 최대값, 최소값 혹은 신호의 기울기가 변하는 지점으로 정의된다. 음성 신호는 저역통과 필터로 먼저 전처리되어 고주파 성분이 제거된다. 이를 통해 불필요한 변곡점들이 제거되며, 피치 주기 추정에 유용한 국소적인 최대값만을 변곡점 검출법을 이용하여 추출한다. 얻어진 변곡점 간의 시간 간격을 측정하여 피치 주기를 추정하며, 그 역수로 기본 주파수 추정치를 얻는다. 기존의 피치 추정 방법은 음성이 국소적으로 시불변이라는 가정하에 음성을 블록 단위로 처리하여 블록당 피치 주기를 구하지만, 제안된 방법은 음성을 샘플 단위로 처리하여 변곡점을 검출하며, 그 결과 피치 주기를 시간 경과에 따라 얻게 되어 음성의 시변성이 반영된 기본 주파수 추정치를 얻는다. 컴퓨터 모의실험으로 기본 주파수 추정기로서 제안된 방법의 유용성을 볼 수 있다.

• 음성과학
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• 제15권1호
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• pp.75-83
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• 2008

The interior of an automobile is a very noisy environment with both stationary cruising noise and the reverberated music or speech coming out from the audio system. For robust speech recognition in a car environment, it is necessary to extract a driver's voice command well by removing those background noises. Since we can handle the music and speech signals from an audio system in a car, the reverberated music and speech sounds can be removed using an acoustic echo canceller. In this paper, we implement an acoustic echo canceller with robust double-talk detection algorithm using TMS-320C6711 DSK. First we developed the echo canceller on the PC for verifying the performance of echo cancellation, then implemented it on the TMS320C6711 DSK. For processing of one speech sample with 8kHz sampling rate and 256 filter taps of the echo canceller, the implemented system used only 0.035ms and achieved the ERLE of 20.73dB.

• 대한음성학회지:말소리
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• 제51호
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• pp.151-165
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• 2004

For educational and research purposes, a Korean speech recognition platform is designed. It is based on an object-oriented architecture and can be easily modified so that researchers can readily evaluate the performance of a recognition algorithm of interest. This platform will save development time for many who are interested in speech recognition. The platform includes the following modules: Noise reduction, end-point detection, met-frequency cepstral coefficient (MFCC) and perceptually linear prediction (PLP)-based feature extraction, hidden Markov model (HMM)-based acoustic modeling, n-gram language modeling, n-best search, and Korean language processing. The decoder of the platform can handle both lexical search trees for large vocabulary speech recognition and finite-state networks for small-to-medium vocabulary speech recognition. It performs word-dependent n-best search algorithm with a bigram language model in the first forward search stage and then extracts a word lattice and restores each lattice path with a trigram language model in the second stage.

• 음성과학
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• 제8권4호
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• pp.35-43
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• 2001

New speech synthesis algorithms capable of flexible prosody (especially F0) modification are desired for a high quality TTS system. TD-PSOLA is the most popular synthesis algorithm. The algorithm shows very high quality when F0 modification is limited. However, the quality degradation due to pitch epoch detection error becomes severe as the F0 modification factor becomes large. On the other hand, the vocoder framework is very flexible in F0 manipulation. The synthesized speech quality from the vocoder is far from natural human speech and suffers from buzziness. To remedy the buzzy quality from the vocoder and make more natural synthetic speech, we propose a mixed phase vocoder.

• 음성과학
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• 제13권3호
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• pp.7-18
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• 2006

Diagnosis of pathological voice is one of the important issues in biomedical applications of speech technology. This study focuses on the discrimination of voice disorder using HMM (Hidden Markov Model) for automatic detection between normal voice and vocal fold disorder voice. This is a non-intrusive, non-expensive and fully automated method using only a speech sample of the subject. Speech data from normal people and patients were collected. Mel-frequency filter cepstral coefficients (MFCCs) were modeled by HMM classifier. Different states (3 states, 5 states and 7 states), 3 mixtures and left to right HMMs were formed. This method gives an accuracy of 93.8% for train data and 91.7% for test data in the discrimination of normal and vocal fold disorder voice for sustained /a/.

• 대한의용생체공학회:의공학회지
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• 제28권5호
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• pp.694-702
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• 2007

The adaptation effect originating from the chemical synapse between auditory nerve and inner hair cell gives advantage in accurate representation of temporal cues of incoming speech such as speech onset. Thus it is expected that the modification of conventional speech processing strategies of cochlear implant(CI) by incorporating the adaptation effect will result in considerable improvement of speech perception performance such as consonant perception score. Our purpose in this paper was to evaluate our new CI speech processing strategy incorporating the adaptation effect by the observation of auditory nerve responses. By classifying the presence or absence of speech from the auditory nerve responses, i. e. spike trains, we could quantitatively compare speech onset detection performances of conventional and improved strategies. We could verify the effectiveness of the adaptation effect in improving the speech onset representation characteristics.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제14권1호
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• pp.168-182
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• 2020

Spoofing attacks, especially replay attacks, pose great security challenges to automatic speaker verification (ASV) systems. Current works on replay attacks detection primarily focused on either developing new features or improving classifier performance, ignoring the effects of feature variability, e.g., the channel variability. In this paper, we first establish a mathematical model for replay speech and introduce a method for eliminating the negative interference of the channel. Then a novel feature is proposed to detect the replay attacks. To further boost the detection performance, four post-processing methods using normalization techniques are investigated. We evaluate our proposed method on the ASVspoof 2017 dataset. The experimental results show that our approach outperforms the competing methods in terms of detection accuracy. More interestingly, we find that the proposed normalization strategy could also improve the performance of the existing algorithms.

• 대한음성학회지:말소리
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• 제66호
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• pp.117-130
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• 2008

This paper presents a method for automatically detecting Korean prosodic boundaries using both acoustic and grammatical information for the performance improvement of speech information processing systems. While most of previous works are solely based on grammatical information, our method utilizes not only grammatical information constructed by a Maximum-Entropy-based grammar model using 10 grammatical features, but also acoustical information constructed by a GMM-based acoustic model using 14 acoustic features. Given that Korean prosodic structure has two intonationally defined prosodic units, intonation phrase (IP) and accentual phrase (AP), experimental results show that the detection rate of AP boundaries is 82.6%, which is higher than the labeler agreement rate in hand transcribing, and that the detection rate of IP boundaries is 88.7%, which is slightly lower than the labeler agreement rate.

• 한국멀티미디어학회논문지
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• 제17권3호
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• pp.294-299
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• 2014

본 논문에서는 일반화된 정규-라플라스(generalized normal-Laplace) 분포 기반의 음성 검출기(voice activity detection) 알고리즘을 제안한다. 제안된 알고리즘은, 잡음 섞인 음성 신호의 확률밀도함수를 일반화된 정규-라플라스 분포로 표현한 다음, 일반화된 정규-라플라스 분포의 음성과 잡음의 분산을 고차 모멘트(higher order moments)를 이용하여 추정한다. 제안된 알고리즘은 다양한 조건의 잡음 환경에서 기존의 음성 검출기들과 비교하였으며 향상된 성능을 보였다.

• ETRI Journal
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• 제31권4호
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• pp.451-453
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• 2009

This letter proposes the use of vowel sound detection for voice activity detection. Vowels have distinctive spectral peaks. These are likely to remain higher than their surroundings even after severe corruption. Therefore, by developing a method of detecting the spectral peaks of vowel sounds in corrupted signals, voice activity can be detected as well even in low signal-to-noise ratio (SNR) conditions. Experimental results indicate that the proposed algorithm performs reliably under various noise and low SNR conditions. This method is suitable for mobile environments where the characteristics of noise may not be known in advance.