• ETRI Journal
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• v.32 no.5
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• pp.831-834
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• 2010

This paper proposes a new joint channel coding algorithm based on principal component analysis. A conventional joint channel coder using passive downmixing undergoes a reduction of both the primary-to-ambient energy ratio (PAR) of the downmix signal and the panning gain ratio of the primary source. The proposed system preserves the PAR of the downmix signal by using active downmixing which reflects spatial characteristic. The proposed system also improves the accuracy of the panning gain ratio estimation. Computer simulations and subjective listening tests verify the performance of the proposed system.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.4
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• pp.118-127
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• 1998

In this paper, we propose a voice personality transformation method which makes one person's voice sound like another person's voice. In order to transform the voice personality, vocal tract transfer function is used as a transformation parameter. Comparing with previous methods, the proposed method can obtain high-quality transformed speech with low computational complexity. Conversion between the vocal tract transfer functions is implemented by a linear mapping based on soft clustering. In this process, mean LPC cepstrum coefficients and mean removed LPC cepstrum modeled by the low dimensional vector are used as transformation parameters. To evaluate the performance of the proposed method, mapping rules are generated from 61 Korean words uttered by two male and one female speakers. These rules are then applied to 9 sentences uttered by the same persons, and objective evaluation and subjective listening tests for the transformed speech are performed.

• Journal of Audiology & Otology
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• v.24 no.4
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• pp.180-190
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• 2020

Background and Objectives: Although many studies have evaluated the effect of the digital noise reduction (DNR) algorithm of hearing aids (HAs) on speech recognition, there are few studies on the effect of DNR on music perception. Therefore, we aimed to evaluate the effect of DNR on music, in addition to speech perception, using objective and subjective measurements. Subjects and Methods: Sixteen HA users participated in this study (58.00±10.44 years; 3 males and 13 females). The objective assessment of speech and music perception was based on the Korean version of the Clinical Assessment of Music Perception test and word and sentence recognition scores. Meanwhile, for the subjective assessment, the quality rating of speech and music as well as self-reported HA benefits were evaluated. Results: There was no improvement conferred with DNR of HAs on the objective assessment tests of speech and music perception. The pitch discrimination at 262 Hz in the DNR-off condition was better than that in the unaided condition (p=0.024); however, the unaided condition and the DNR-on conditions did not differ. In the Korean music background questionnaire, responses regarding ease of communication were better in the DNR-on condition than in the DNR-off condition (p=0.029). Conclusions: Speech and music perception or sound quality did not improve with the activation of DNR. However, DNR positively influenced the listener's subjective listening comfort. The DNR-off condition in HAs may be beneficial for pitch discrimination at some frequencies.

• Korean Journal of Audiology
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• v.24 no.4
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• pp.180-190
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• 2020

Background and Objectives: Although many studies have evaluated the effect of the digital noise reduction (DNR) algorithm of hearing aids (HAs) on speech recognition, there are few studies on the effect of DNR on music perception. Therefore, we aimed to evaluate the effect of DNR on music, in addition to speech perception, using objective and subjective measurements. Subjects and Methods: Sixteen HA users participated in this study (58.00±10.44 years; 3 males and 13 females). The objective assessment of speech and music perception was based on the Korean version of the Clinical Assessment of Music Perception test and word and sentence recognition scores. Meanwhile, for the subjective assessment, the quality rating of speech and music as well as self-reported HA benefits were evaluated. Results: There was no improvement conferred with DNR of HAs on the objective assessment tests of speech and music perception. The pitch discrimination at 262 Hz in the DNR-off condition was better than that in the unaided condition (p=0.024); however, the unaided condition and the DNR-on conditions did not differ. In the Korean music background questionnaire, responses regarding ease of communication were better in the DNR-on condition than in the DNR-off condition (p=0.029). Conclusions: Speech and music perception or sound quality did not improve with the activation of DNR. However, DNR positively influenced the listener's subjective listening comfort. The DNR-off condition in HAs may be beneficial for pitch discrimination at some frequencies.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.58-65
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• 2001

Recently, a 5-degrees-of-freedom (DOF) reverberation model was proposed as a method of representing subjective perception of reverberation as objective measures[1]. This model approximates sound energy decay curve by five objective measures, widely used in which have been concert hall acoustics. However, it is note worthy that there can be infinite number of impulse responses which correspond to a selected 5-DOF reverberation model. There may exist some filters making very unnatural and unrealistic sound. In this paper, the limitation of the 5-DOF reverberation model when it is used as a filter design criteria is investigated. When a 5-DOF reverberation model is given, additional constraints to get natural reverberation are suggested. This is based on the listening tests for several quite different source sounds.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.22-32
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• 2011

The present study examined the effects of the complexity of the 3D models on the results of acoustic simulation which is the predominant tool of the acoustical design of buildings. Also, the effects of the 3D model on the auralized sounds were investigated. In order to carry out the study, four 3D models with different levels of complexity were introduced for a real auditorium which have different numbers of surfaces in the persuit of the guidance of odeon room acoustic software. The set-up of models was also based on the level of transition order of the program. And the acoustic experiments were performed measuring room acoustic parameters including SPL, RT, C80, D50. Acoustic computer simulations were performed using four different models. Then, the results of the computer modeling were compared with the measured acoustical parameters. In addition, sound sources were recorded in the field and auralized sounds were made in convolution with the impulse source made from acoustic modeling. Then, subjective tests were undertaken using auralized sounds. As the results, it was found that the result of the acoustic simulation were closer to the real room acoustic properties when 3D model was more particularly made. For the subjective test, the listening materials were acknowledged as similar with the real sound source when more complex 3D model was used. Then, it could be concluded that the complexity of the 3D model affects the results of the acoustic modeling as well as subjective tests.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.184-190
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• 2013

Parametric stereo(PS) audio coding is a specific version of spatial audio coding. In this paper, the problem due to the conventional synthesis of phase differences. In the conventional upmix matrix, phase differences are synthesized not only on downmix signal but also ambient signal, which violates the assumption that the ambient signals are anti-phased. Deterioration due to the phase synthesis is analyzed, especially, for low interchannel correlation. To solve this problem, new upmix matrix is proposed, which synthesizes phase differences only on downmix signal. The performance of the proposed upmix matrix is verified by the subjective listening tests.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.1430-1436
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• 2007

A principal components analysis (PCA) of the median head-related impulse responses (HRIRs) in the CIPIC HRTF database reveals that the individual HRIRs can be adequately reconstructed by a linear combination of 12 orthonormal basis functions. These basis functions can be used generally to model arbitrary HRIRs, which are not included in the process to obtain the basis functions. To clarify whether these basis functions can be used to model other set of arbitrary HRIRs, an numerical error analysis for modeling and a series of subjective listening tests were carried out using the measured and modeled HRIRs. The results showed that the set of individual HRIRs, which were measured in our lab using different measurement conditions, techniques, and source positions, can be well modeled with reasonable accuracy. Furthermore, all subjects reported not only the accurate vertical perception but also the front-back discrimination with the modeled HRIRs based on 12 basis functions. However, as less basis functions were used for HRIR modeling, the modeling accuracy and localization performance deteriorated.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.3
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• pp.139-145
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• 2013

In this paper, the problems of the conventional amplitude panning method for a stereophonic panning system are analyzed. We observed that the distortion showed a feedforward comb filter response. As a remedy to this distortion, we propose a stereophonic panning system using a feedback comb filter. The comb filter is designed to minimize the difference between interaural level difference(ILD) of the proposed system and that of HRTF because ILD is most salient cue for the perception of the source direction. The proposed system is configured to operate selectively for the frequency band related to the source direction. The performance of the proposed system is verified by subjective listening tests.

• Journal of Korea Multimedia Society
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• v.4 no.1
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• pp.56-66
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• 2001

The spectral subtraction algorithm using a criterion based on the human perception has been recently developed. The speech processed with Virag's algorithm sounds more pleasant to a human listener than those obtained by the classical methods. However, Virag's algorithm requires a robust voice activity detector (VAD). In the ESS (extended spectral subtraction) algorithm without VAD, the residual noise becomes more noticeable as the SNR decrease. In this paper we propose a new speech enhancement method, the combination of Wiener filter and spectral subtraction based on noise masking characteristics in the human auditory system. There is no need of VAD because the noise can be successively updated even during speech activity using Wiener filter. The adjustment of the subtraction parameter based on the masking threshold makes the residual noise inaudible. The proposed method has been compared with conventional spectral subtraction algorithms. Objective and subjective evaluation of the proposed system is performed with several noise types having different time-frequency distributions. The application of objective measures, the study of the speech spectrograms, as well as subjective listening tests, confirm that the enhanced speech with proposed algorithm is more pleasant to a human listener.