• Title/Summary/Keyword: Acoustic Feature

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Performance assessments of feature vectors and classification algorithms for amphibian sound classification (양서류 울음 소리 식별을 위한 특징 벡터 및 인식 알고리즘 성능 분석)

  • Park, Sangwook;Ko, Kyungdeuk;Ko, Hanseok
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.6
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    • pp.401-406
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    • 2017
  • This paper presents the performance assessment of several key algorithms conducted for amphibian species sound classification. Firstly, 9 target species including endangered species are defined and a database of their sounds is built. For performance assessment, three feature vectors such as MFCC (Mel Frequency Cepstral Coefficient), RCGCC (Robust Compressive Gammachirp filterbank Cepstral Coefficient), and SPCC (Subspace Projection Cepstral Coefficient), and three classifiers such as GMM(Gaussian Mixture Model), SVM(Support Vector Machine), DBN-DNN(Deep Belief Network - Deep Neural Network) are considered. In addition, i-vector based classification system which is widely used for speaker recognition, is used to assess for this task. Experimental results indicate that, SPCC-SVM achieved the best performance with 98.81 % while other methods also attained good performance with above 90 %.

A New Temporal Filtering Method for Improved Automatic Lipreading (향상된 자동 독순을 위한 새로운 시간영역 필터링 기법)

  • Lee, Jong-Seok;Park, Cheol-Hoon
    • The KIPS Transactions:PartB
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    • v.15B no.2
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    • pp.123-130
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    • 2008
  • Automatic lipreading is to recognize speech by observing the movement of a speaker's lips. It has received attention recently as a method of complementing performance degradation of acoustic speech recognition in acoustically noisy environments. One of the important issues in automatic lipreading is to define and extract salient features from the recorded images. In this paper, we propose a feature extraction method by using a new filtering technique for obtaining improved recognition performance. The proposed method eliminates frequency components which are too slow or too fast compared to the relevant speech information by applying a band-pass filter to the temporal trajectory of each pixel in the images containing the lip region and, then, features are extracted by principal component analysis. We show that the proposed method produces improved performance in both clean and visually noisy conditions via speaker-independent recognition experiments.

CNN-based Automatic Machine Fault Diagnosis Method Using Spectrogram Images (스펙트로그램 이미지를 이용한 CNN 기반 자동화 기계 고장 진단 기법)

  • Kang, Kyung-Won;Lee, Kyeong-Min
    • Journal of the Institute of Convergence Signal Processing
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    • v.21 no.3
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    • pp.121-126
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    • 2020
  • Sound-based machine fault diagnosis is the automatic detection of abnormal sound in the acoustic emission signals of the machines. Conventional methods of using mathematical models were difficult to diagnose machine failure due to the complexity of the industry machinery system and the existence of nonlinear factors such as noises. Therefore, we want to solve the problem of machine fault diagnosis as a deep learning-based image classification problem. In the paper, we propose a CNN-based automatic machine fault diagnosis method using Spectrogram images. The proposed method uses STFT to effectively extract feature vectors from frequencies generated by machine defects, and the feature vectors detected by STFT were converted into spectrogram images and classified by CNN by machine status. The results show that the proposed method can be effectively used not only to detect defects but also to various automatic diagnosis system based on sound.

Hierarchical Flow-Based Anomaly Detection Model for Motor Gearbox Defect Detection

  • Younghwa Lee;Il-Sik Chang;Suseong Oh;Youngjin Nam;Youngteuk Chae;Geonyoung Choi;Gooman Park
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.17 no.6
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    • pp.1516-1529
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    • 2023
  • In this paper, a motor gearbox fault-detection system based on a hierarchical flow-based model is proposed. The proposed system is used for the anomaly detection of a motion sound-based actuator module. The proposed flow-based model, which is a generative model, learns by directly modeling a data distribution function. As the objective function is the maximum likelihood value of the input data, the training is stable and simple to use for anomaly detection. The operation sound of a car's side-view mirror motor is converted into a Mel-spectrogram image, consisting of a folding signal and an unfolding signal, and used as training data in this experiment. The proposed system is composed of an encoder and a decoder. The data extracted from the layer of the pretrained feature extractor are used as the decoder input data in the encoder. This information is used in the decoder by performing an interlayer cross-scale convolution operation. The experimental results indicate that the context information of various dimensions extracted from the interlayer hierarchical data improves the defect detection accuracy. This paper is notable because it uses acoustic data and a normalizing flow model to detect outliers based on the features of experimental data.

Dialect classification based on the speed and the pause of speech utterances (발화 속도와 휴지 구간 길이를 사용한 방언 분류)

  • Jonghwan Na;Bowon Lee
    • Phonetics and Speech Sciences
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    • v.15 no.2
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    • pp.43-51
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    • 2023
  • In this paper, we propose an approach for dialect classification based on the speed and pause of speech utterances as well as the age and gender of the speakers. Dialect classification is one of the important techniques for speech analysis. For example, an accurate dialect classification model can potentially improve the performance of speaker or speech recognition. According to previous studies, research based on deep learning using Mel-Frequency Cepstral Coefficients (MFCC) features has been the dominant approach. We focus on the acoustic differences between regions and conduct dialect classification based on the extracted features derived from the differences. In this paper, we propose an approach of extracting underexplored additional features, namely the speed and the pauses of speech utterances along with the metadata including the age and the gender of the speakers. Experimental results show that our proposed approach results in higher accuracy, especially with the speech rate feature, compared to the method only using the MFCC features. The accuracy improved from 91.02% to 97.02% compared to the previous method that only used MFCC features, by incorporating all the proposed features in this paper.

Frequency Domain Double-Talk Detector Based on Gaussian Mixture Model (주파수 영역에서의 Gaussian Mixture Model 기반의 동시통화 검출 연구)

  • Lee, Kyu-Ho;Chang, Joon-Hyuk
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.4
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    • pp.401-407
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    • 2009
  • In this paper, we propose a novel method for the cross-correlation based double-talk detection (DTD), which employing the Gaussian Mixture Model (GMM) in the frequency domain. The proposed algorithm transforms the cross correlation coefficient used in the time domain into 16 channels in the frequency domain using the discrete fourier transform (DFT). The channels are then selected into seven feature vectors for GMM and we identify three different regions such as far-end, double-talk and near-end speech using the likelihood comparison based on those feature vectors. The presented DTD algorithm detects efficiently the double-talk regions without Voice Activity Detector which has been used in conventional cross correlation based double-talk detection. The performance of the proposed algorithm is evaluated under various conditions and yields better results compared with the conventional schemes. especially, show the robustness against detection errors resulting from the background noises or echo path change which one of the key issues in practical DTD.

A pilot study on the application of environmental DNA to the estimation of the biomass of dominant species in the northwestern waters of Jeju Island (제주도 서북 해역에서의 우점종 생물량 추정에 환경 유전자의 적용에 관한 시범 연구)

  • KANG, Myounghee;PARK, Kyeong-Dong;MIN, Eunbi;LEE, Changheon;KANG, Taejong;OH, Taegeon;LIM, Byeonggwon;HWANG, Doojin;KIM, Byung-Yeob
    • Journal of the Korean Society of Fisheries and Ocean Technology
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    • v.58 no.1
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    • pp.39-48
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    • 2022
  • Using environmental DNA (eDNA) in the fisheries and oceanography fields, research on the diversity of biological species, the presence or absence of specific species and quantitative evaluation of species has considerably been performed. Up to date, no study on eDNA has been tried in the area of fisheries acoustics in Korea. In this study, the biomass of a dominant species in the northwestern waters of Jeju Island was examined using 1) the catch ratio of the species from trawl survey results and 2) the ranking ratio of the species from the eDNA results. The dominant species was Zoarces gillii, and its trawl catch ratio was 68.2% and its eDNA ratio was 81.3%. The Zoarces gillii biomass from the two methods was 7199.4 tons (trawl) and 8584.6 tons (eDNA), respectively. The mean and standard deviation of the acoustic backscattering strength values (120 kHz) from the entire survey area were 135.5 and 157.7 m2/nm2, respectively. The strongest echo signal occurred at latitude 34° and longitude 126°15' (northwest of Jeju Island). High echo signals were observed in a specific oceanographic feature (salinity range of 32-33 psu and the water temperature range of 19-20℃). This study was a pilot study on evaluating quantitatively aquatic resources by applying the eDNA technique into acoustic-trawl survey method. Points to be considered for high-quality quantitative estimation using the eDNA to fisheries acosutics were discussed.

Robust Person Identification Using Optimal Reliability in Audio-Visual Information Fusion

  • Tariquzzaman, Md.;Kim, Jin-Young;Na, Seung-You;Choi, Seung-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.3E
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    • pp.109-117
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    • 2009
  • Identity recognition in real environment with a reliable mode is a key issue in human computer interaction (HCI). In this paper, we present a robust person identification system considering score-based optimal reliability measure of audio-visual modalities. We propose an extension of the modified reliability function by introducing optimizing parameters for both of audio and visual modalities. For degradation of visual signals, we have applied JPEG compression to test images. In addition, for creating mismatch in between enrollment and test session, acoustic Babble noises and artificial illumination have been added to test audio and visual signals, respectively. Local PCA has been used on both modalities to reduce the dimension of feature vector. We have applied a swarm intelligence algorithm, i.e., particle swarm optimization for optimizing the modified convection function's optimizing parameters. The overall person identification experiments are performed using VidTimit DB. Experimental results show that our proposed optimal reliability measures have effectively enhanced the identification accuracy of 7.73% and 8.18% at different illumination direction to visual signal and consequent Babble noises to audio signal, respectively, in comparison with the best classifier system in the fusion system and maintained the modality reliability statistics in terms of its performance; it thus verified the consistency of the proposed extension.

An Amplitude Warping Approach to Intra-Speaker Normalization for Speech Recognition (음성인식에서 화자 내 정규화를 위한 진폭 변경 방법)

  • Kim Dong-Hyun;Hong Kwang-Seok
    • Journal of Internet Computing and Services
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    • v.4 no.3
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    • pp.9-14
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    • 2003
  • The method of vocal tract normalization is a successful method for improving the accuracy of inter-speaker normalization. In this paper, we present an intra-speaker warping factor estimation based on pitch alteration utterance. The feature space distributions of untransformed speech from the pitch alteration utterance of intra-speaker would vary due to the acoustic differences of speech produced by glottis and vocal tract. The variation of utterance is two types: frequency and amplitude variation. The vocal tract normalization is frequency normalization among inter-speaker normalization methods. Therefore, we have to consider amplitude variation, and it may be possible to determine the amplitude warping factor by calculating the inverse ratio of input to reference pitch. k, the recognition results, the error rate is reduced from 0.4% to 2.3% for digit and word decoding.

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Automatic pronunciation assessment of English produced by Korean learners using articulatory features (조음자질을 이용한 한국인 학습자의 영어 발화 자동 발음 평가)

  • Ryu, Hyuksu;Chung, Minhwa
    • Phonetics and Speech Sciences
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    • v.8 no.4
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    • pp.103-113
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    • 2016
  • This paper aims to propose articulatory features as novel predictors for automatic pronunciation assessment of English produced by Korean learners. Based on the distinctive feature theory, where phonemes are represented as a set of articulatory/phonetic properties, we propose articulatory Goodness-Of-Pronunciation(aGOP) features in terms of the corresponding articulatory attributes, such as nasal, sonorant, anterior, etc. An English speech corpus spoken by Korean learners is used in the assessment modeling. In our system, learners' speech is forced aligned and recognized by using the acoustic and pronunciation models derived from the WSJ corpus (native North American speech) and the CMU pronouncing dictionary, respectively. In order to compute aGOP features, articulatory models are trained for the corresponding articulatory attributes. In addition to the proposed features, various features which are divided into four categories such as RATE, SEGMENT, SILENCE, and GOP are applied as a baseline. In order to enhance the assessment modeling performance and investigate the weights of the salient features, relevant features are extracted by using Best Subset Selection(BSS). The results show that the proposed model using aGOP features outperform the baseline. In addition, analysis of relevant features extracted by BSS reveals that the selected aGOP features represent the salient variations of Korean learners of English. The results are expected to be effective for automatic pronunciation error detection, as well.