• Title/Summary/Keyword: Mel frequency cepstral coefficients

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Speech/Music Discrimination Using Mel-Cepstrum Modulation Energy (멜 켑스트럼 모듈레이션 에너지를 이용한 음성/음악 판별)

  • Kim, Bong-Wan;Choi, Dea-Lim;Lee, Yong-Ju
    • MALSORI
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    • no.64
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    • pp.89-103
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    • 2007
  • In this paper, we introduce mel-cepstrum modulation energy (MCME) for a feature to discriminate speech and music data. MCME is a mel-cepstrum domain extension of modulation energy (ME). MCME is extracted on the time trajectory of Mel-frequency cepstral coefficients, while ME is based on the spectrum. As cepstral coefficients are mutually uncorrelated, we expect the MCME to perform better than the ME. To find out the best modulation frequency for MCME, we perform experiments with 4 Hz to 20 Hz modulation frequency. To show effectiveness of the proposed feature, MCME, we compare the discrimination accuracy with the results obtained from the ME and the cepstral flux.

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Noise Robust Text-Independent Speaker Identification for Ubiquitous Robot Companion (지능형 서비스 로봇을 위한 잡음에 강인한 문맥독립 화자식별 시스템)

  • Kim, Sung-Tak;Ji, Mi-Kyoung;Kim, Hoi-Rin;Kim, Hye-Jin;Yoon, Ho-Sub
    • 한국HCI학회:학술대회논문집
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    • 2008.02a
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    • pp.190-194
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    • 2008
  • This paper presents a speaker identification technique which is one of the basic techniques of the ubiquitous robot companion. Though the conventional mel-frequency cepstral coefficients guarantee high performance of speaker identification in clean condition, the performance is degraded dramatically in noise condition. To overcome this problem, we employed the relative autocorrelation sequence mel-frequency cepstral coefficient which is one of the noise robust features. However, there are two problems in relative autocorrelation sequence mel-frequency cepstral coefficient: 1) the limited information problem. 2) the residual noise problem. In this paper, to deal with these drawbacks, we propose a multi-streaming method for the limited information problem and a hybrid method for the residual noise problem. To evaluate proposed methods, noisy speech is used in which air conditioner noise, classic music, and vacuum noise are artificially added. Through experiments, proposed methods provide better performance of speaker identification than the conventional methods.

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Sound Reinforcement Based on Context Awareness for Hearing Impaired (청각장애인을 위한 상황인지기반의 음향강화기술)

  • Choi, Jae-Hun;Chang, Joon-Hyuk
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.48 no.5
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    • pp.109-114
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    • 2011
  • In this paper, we apply a context awareness based on Gaussian mixture model (GMM) to a sound reinforcement for hearing impaired. In our approach, the harmful sound amplified through the sound reinforcement algorithm according to context awareness based on GMM which is constructed as Mel-frequency cepstral coefficients (MFCC) feature vector from sound data. According to the experimental results, the proposed approach is found to be effective in the various acoustic environments.

Mel-Frequency Cepstral Coefficients Using Formants-Based Gaussian Distribution Filterbank (포만트 기반의 가우시안 분포를 가지는 필터뱅크를 이용한 멜-주파수 켑스트럴 계수)

  • Son, Young-Woo;Hong, Jae-Keun
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.8
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    • pp.370-374
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    • 2006
  • Mel-frequency cepstral coefficients are widely used as the feature for speech recognition. In FMCC extraction process. the spectrum. obtained by Fourier transform of input speech signal is divided by met-frequency bands, and each band energy is extracted for the each frequency band. The coefficients are extracted by the discrete cosine transform of the obtained band energy. In this Paper. we calculate the output energy for each bandpass filter by taking the weighting function when applying met-frequency scaled bandpass filter. The weighting function is Gaussian distributed function whose center is at the formant frequency In the experiments, we can see the comparative performance with the standard MFCC in clean condition. and the better Performance in worse condition by the method proposed here.

A New Feature for Speech Segments Extraction with Hidden Markov Models (숨은마코프모형을 이용하는 음성구간 추출을 위한 특징벡터)

  • Hong, Jeong-Woo;Oh, Chang-Hyuck
    • Communications for Statistical Applications and Methods
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    • v.15 no.2
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    • pp.293-302
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    • 2008
  • In this paper we propose a new feature, average power, for speech segments extraction with hidden Markov models, which is based on mel frequencies of speech signals. The average power is compared with the mel frequency cepstral coefficients, MFCC, and the power coefficient. To compare performances of three types of features, speech data are collected for words with explosives which are generally known hard to be detected. Experiments show that the average power is more accurate and efficient than MFCC and the power coefficient for speech segments extraction in environments with various levels of noise.

Evaluation of Frequency Warping Based Features and Spectro-Temporal Features for Speaker Recognition (화자인식을 위한 주파수 워핑 기반 특징 및 주파수-시간 특징 평가)

  • Choi, Young Ho;Ban, Sung Min;Kim, Kyung-Wha;Kim, Hyung Soon
    • Phonetics and Speech Sciences
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    • v.7 no.1
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    • pp.3-10
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    • 2015
  • In this paper, different frequency scales in cepstral feature extraction are evaluated for the text-independent speaker recognition. To this end, mel-frequency cepstral coefficients (MFCCs), linear frequency cepstral coefficients (LFCCs), and bilinear warped frequency cepstral coefficients (BWFCCs) are applied to the speaker recognition experiment. In addition, the spectro-temporal features extracted by the cepstral-time matrix (CTM) are examined as an alternative to the delta and delta-delta features. Experiments on the NIST speaker recognition evaluation (SRE) 2004 task are carried out using the Gaussian mixture model-universal background model (GMM-UBM) method and the joint factor analysis (JFA) method, both based on the ALIZE 3.0 toolkit. Experimental results using both the methods show that BWFCC with appropriate warping factor yields better performance than MFCC and LFCC. It is also shown that the feature set including the spectro-temporal information based on the CTM outperforms the conventional feature set including the delta and delta-delta features.

A Study on Stable Motion Control of Humanoid Robot with 24 Joints Based on Voice Command

  • Lee, Woo-Song;Kim, Min-Seong;Bae, Ho-Young;Jung, Yang-Keun;Jung, Young-Hwa;Shin, Gi-Soo;Park, In-Man;Han, Sung-Hyun
    • Journal of the Korean Society of Industry Convergence
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    • v.21 no.1
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    • pp.17-27
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    • 2018
  • We propose a new approach to control a biped robot motion based on iterative learning of voice command for the implementation of smart factory. The real-time processing of speech signal is very important for high-speed and precise automatic voice recognition technology. Recently, voice recognition is being used for intelligent robot control, artificial life, wireless communication and IoT application. In order to extract valuable information from the speech signal, make decisions on the process, and obtain results, the data needs to be manipulated and analyzed. Basic method used for extracting the features of the voice signal is to find the Mel frequency cepstral coefficients. Mel-frequency cepstral coefficients are the coefficients that collectively represent the short-term power spectrum of a sound, based on a linear cosine transform of a log power spectrum on a nonlinear mel scale of frequency. The reliability of voice command to control of the biped robot's motion is illustrated by computer simulation and experiment for biped walking robot with 24 joint.

Digital Isolated Word Recognition System based on MFCC and DTW Algorithm (MFCC와 DTW에 알고리즘을 기반으로 한 디지털 고립단어 인식 시스템)

  • Zang, Xian;Chong, Kil-To
    • Proceedings of the KIEE Conference
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    • 2008.10b
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    • pp.290-291
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    • 2008
  • The most popular speech feature used in speech recognition today is the Mel-Frequency Cepstral Coefficients (MFCC) algorithm, which could reflect the perception characteristics of the human ear more accurately than other parameters. This paper adopts MFCC and its first order difference, which could reflect the dynamic character of speech signal, as synthetical parametric representation. Furthermore, we quote Dynamic Time Warping (DTW) algorithm to search match paths in the pattern recognition process. We use the software "GoldWave" to record English digitals in the lab environments and the simulation results indicate the algorithm has higher recognition accuracy than others using LPCC, etc. as character parameters in the experiment for Digital Isolated Word Recognition (DIWR) system.

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Discriminative Weight Training for Gender Identification (변별적 가중치 학습을 적용한 성별인식 알고리즘)

  • Kang, Sang-Ick;Chang, Joon-Hyuk
    • The Journal of the Acoustical Society of Korea
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    • v.27 no.5
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    • pp.252-255
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    • 2008
  • In this paper, we apply a discriminative weight training to a support vector machine (SVM) based gender identification. In our approach, the gender decision rule is expressed as the SVM of optimally weighted mel-frequency cepstral coefficients (MFCC) based on a minimum classification error (MCE) method which is different from the previous works in that different weights are assigned to each MFCC filter bank which is considered more realistic. According to the experimental results, the proposed approach is found to be effective for gender identification using SVM.

Speech Emotion Recognition Using 2D-CNN with Mel-Frequency Cepstrum Coefficients

  • Eom, Youngsik;Bang, Junseong
    • Journal of information and communication convergence engineering
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    • v.19 no.3
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    • pp.148-154
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    • 2021
  • With the advent of context-aware computing, many attempts were made to understand emotions. Among these various attempts, Speech Emotion Recognition (SER) is a method of recognizing the speaker's emotions through speech information. The SER is successful in selecting distinctive 'features' and 'classifying' them in an appropriate way. In this paper, the performances of SER using neural network models (e.g., fully connected network (FCN), convolutional neural network (CNN)) with Mel-Frequency Cepstral Coefficients (MFCC) are examined in terms of the accuracy and distribution of emotion recognition. For Ryerson Audio-Visual Database of Emotional Speech and Song (RAVDESS) dataset, by tuning model parameters, a two-dimensional Convolutional Neural Network (2D-CNN) model with MFCC showed the best performance with an average accuracy of 88.54% for 5 emotions, anger, happiness, calm, fear, and sadness, of men and women. In addition, by examining the distribution of emotion recognition accuracies for neural network models, the 2D-CNN with MFCC can expect an overall accuracy of 75% or more.