• Title/Summary/Keyword: perceptual linear prediction

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Glottal Weighted Cepstrum for Robust Speech Recognition (잡음에 강한 음성 인식을 위한 성문 가중 켑스트럼에 관한 연구)

  • 전선도;강철호
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.5
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    • pp.78-82
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    • 1999
  • This paper is a study on weighted cepstrum used broadly for robust speech recognition. Especially, we propose the weighted function of asymmetric glottal pulse shape. which is used for weighted cepstrum extracted by PLP(Perceptual Linear Predictive) based on auditory model. Also, we analyze this glottal weighted cepstrum from the glottal pulse of glottal model in connection with the cepstrum. And we obtain speech features analyzed by both the glottal model and the auditory model. The isolated-word recognition rate is adopted for the test of proposed method in the car moise and street environment. And the performance of glottal weighted cepstrum is compared with both that of weighted cepstrum extracted by LP(Linear Prediction) and that of weighted cepstrum extracted by PLP. The result of computer simulation shows that recognition rate of the proposed glottal weighted cepstrum is better than those of other weighted cepstrums.

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The Speech Recognition Method by Perceptual Linear Predictive Analysis (인지 선형 예측 분석에 의한 음성 인식 방법)

  • 김현철
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1995.06a
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    • pp.184-187
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    • 1995
  • This paper proposes an algorithm for machine recognition of phonemes in continuous speech. The proposed algorithm is static strategy neural network. The algorithm uses, at the stage of training neuron, features such as PARCOR coefficient and auditory-like perceptual liner prediction . These features are extracted from speech samples selected by a sliding 25.6msec windows with s sliding gap being 3 msec long, then interleaved and summed up to 7 sets of parmeters covering 171 msec worth of speech for use of neural inputs. Perfomances are compared when either PARCOR or auditory-like PLP is included in the feture set.

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Multi Mode Harmonic Transform Coding for Speech and Music

  • Kim, Jonghark;Shin, Jae-Hyun;Lee, Insung
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.3E
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    • pp.101-109
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    • 2003
  • A multi-mode harmonic transform coding (MMHTC) for speech and music signals is proposed. Its structure is organized as a linear prediction model with an input of harmonic and transform-based excitation. The proposed coder also utilizes harmonic prediction and an improved quantizer of excitation signal. To efficiently quantize the excitation of music signals, the modulated lapped transform(MLT) is introduced. In other words, the coder combines both the time domain (linear prediction) and the frequency domain technique to achieve the best perceptual quality. The proposed coder showed better speech quality than that of the 8 kbps QCELP coder at a bit-rate of 4 kbps.

Speech Recognition Using Noise Robust Features and Spectral Subtraction (잡음에 강한 특징 벡터 및 스펙트럼 차감법을 이용한 음성 인식)

  • Shin, Won-Ho;Yang, Tae-Young;Kim, Weon-Goo;Youn, Dae-Hee;Seo, Young-Joo
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.5
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    • pp.38-43
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    • 1996
  • This paper compares the recognition performances of feature vectors known to be robust to the environmental noise. And, the speech subtraction technique is combined with the noise robust feature to get more performance enhancement. The experiments using SMC(Short time Modified Coherence) analysis, root cepstral analysis, LDA(Linear Discriminant Analysis), PLP(Perceptual Linear Prediction), RASTA(RelAtive SpecTrAl) processing are carried out. An isolated word recognition system is composed using semi-continuous HMM. Noisy environment experiments usign two types of noises:exhibition hall, computer room are carried out at 0, 10, 20dB SNRs. The experimental result shows that SMC and root based mel cepstrum(root_mel cepstrum) show 9.86% and 12.68% recognition enhancement at 10dB in compare to the LPCC(Linear Prediction Cepstral Coefficient). And when combined with spectral subtraction, mel cepstrum and root_mel cepstrum show 16.7% and 8.4% enhanced recognition rate of 94.91% and 94.28% at 10dB.

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Performance Comparison of Deep Feature Based Speaker Verification Systems (깊은 신경망 특징 기반 화자 검증 시스템의 성능 비교)

  • Kim, Dae Hyun;Seong, Woo Kyeong;Kim, Hong Kook
    • Phonetics and Speech Sciences
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    • v.7 no.4
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    • pp.9-16
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    • 2015
  • In this paper, several experiments are performed according to deep neural network (DNN) based features for the performance comparison of speaker verification (SV) systems. To this end, input features for a DNN, such as mel-frequency cepstral coefficient (MFCC), linear-frequency cepstral coefficient (LFCC), and perceptual linear prediction (PLP), are first compared in a view of the SV performance. After that, the effect of a DNN training method and a structure of hidden layers of DNNs on the SV performance is investigated depending on the type of features. The performance of an SV system is then evaluated on the basis of I-vector or probabilistic linear discriminant analysis (PLDA) scoring method. It is shown from SV experiments that a tandem feature of DNN bottleneck feature and MFCC feature gives the best performance when DNNs are configured using a rectangular type of hidden layers and trained with a supervised training method.

Efficient Harmonic-CELP Based Low Bit Rate Speech Coder (효율적인 하모닉-CELP 구조를 갖는 저 전송률 음성 부호화기)

  • 최용수;김경민;윤대희
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.5
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    • pp.35-47
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    • 2001
  • This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

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An Emotion Recognition Technique using Speech Signals (음성신호를 이용한 감정인식)

  • Jung, Byung-Wook;Cheun, Seung-Pyo;Kim, Youn-Tae;Kim, Sung-Shin
    • Journal of the Korean Institute of Intelligent Systems
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    • v.18 no.4
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    • pp.494-500
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    • 2008
  • In the field of development of human interface technology, the interactions between human and machine are important. The research on emotion recognition helps these interactions. This paper presents an algorithm for emotion recognition based on personalized speech signals. The proposed approach is trying to extract the characteristic of speech signal for emotion recognition using PLP (perceptual linear prediction) analysis. The PLP analysis technique was originally designed to suppress speaker dependent components in features used for automatic speech recognition, but later experiments demonstrated the efficiency of their use for speaker recognition tasks. So this paper proposed an algorithm that can easily evaluate the personal emotion from speech signals in real time using personalized emotion patterns that are made by PLP analysis. The experimental results show that the maximum recognition rate for the speaker dependant system is above 90%, whereas the average recognition rate is 75%. The proposed system has a simple structure and but efficient to be used in real time.

A Method of Evaluating Korean Articulation Quality for Rehabilitation of Articulation Disorder in Children

  • Lee, Keonsoo;Nam, Yunyoung
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.14 no.8
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    • pp.3257-3269
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    • 2020
  • Articulation disorders are characterized by an inability to achieve clear pronunciation due to misuse of the articulators. In this paper, a method of detecting such disorders by comparing to the standard pronunciations is proposed. This method defines the standard pronunciations from the speeches of normal children by clustering them with three features which are the Linear Predictive Cepstral Coefficient (LPCC), the Mel-Frequency Cepstral Coefficient (MFCC), and the Relative Spectral Analysis Perceptual Linear Prediction (RASTA-PLP). By calculating the distance between the centroid of the standard pronunciation and the inputted pronunciation, disordered speech whose features locates outside the cluster is detected. 89 children (58 of normal children and 31 of children with disorders) were recruited. 35 U-TAP test words were selected and each word's standard pronunciation is made from normal children and compared to each pronunciation of children with disorders. In the experiments, the pronunciations with disorders were successfully distinguished from the standard pronunciations.

Quality Improvement of Low Bitrate HE-AAC using Linear Prediction Pre-processor (저 전송률 환경에서 선형예측 전처리기를 사용한 HE-AAC의 성능 향상)

  • Lee, Jae-Seong;Lee, Gun-Woo;Park, Young-Chul;Youn, Dae-Hee
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.8C
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    • pp.822-829
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    • 2009
  • This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

An Objective No-Reference Perceptual Quality Assessment Metric based on Temporal Complexity and Disparity for Stereoscopic Video

  • Ha, Kwangsung;Bae, Sung-Ho;Kim, Munchurl
    • IEIE Transactions on Smart Processing and Computing
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    • v.2 no.5
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    • pp.255-265
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    • 2013
  • 3DTV is expected to be a promising next-generation broadcasting service. On the other hand, the visual discomfort/fatigue problems caused by viewing 3D videos have become an important issue. This paper proposes a perceptual quality assessment metric for a stereoscopic video (SV-PQAM). To model the SV-PQAM, this paper presents the following features: temporal variance, disparity variation in intra-frames, disparity variation in inter-frames and disparity distribution of frame boundary areas, which affect the human perception of depth and visual discomfort for stereoscopic views. The four features were combined into the SV-PQAM, which then becomes a no-reference stereoscopic video quality perception model, as an objective quality assessment metric. The proposed SV-PQAM does not require a depth map but instead uses the disparity information by a simple estimation. The model parameters were estimated based on linear regression from the mean score opinion values obtained from the subjective perception quality assessments. The experimental results showed that the proposed SV-PQAM exhibits high consistency with subjective perception quality assessment results in terms of the Pearson correlation coefficient value of 0.808, and the prediction performance exhibited good consistency with a zero outlier ratio value.

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