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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of the Acoustical Society of Korea
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Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 23, Issue 8 - Nov 2004
Volume 23, Issue 7 - Oct 2004
Volume 23, Issue 6 - Aug 2004
Volume 23, Issue 5 - Jul 2004
Volume 23, Issue 4 - May 2004
Volume 23, Issue 3 - Apr 2004
Volume 23, Issue 2 - Feb 2004
Volume 23, Issue 1 - Jan 2004
Volume 23, Issue 4E - 00 2004
Volume 23, Issue 3E - 00 2004
Volume 23, Issue 2E - 00 2004
Volume 23, Issue 1_E - 00 2004
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An Experimental Study on Synthetic Aperture Sonar under Korean Littoral Environment
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 428~436
Synthetic Aperture Sonar is a technique of extending Physically limited length of an array by signal processing to enhance bearing resolution of a system. The previous techniques estimate most or away shapes as linear. so when towed array shapes are distorted. this can create a deviation from actual situation. In this paper. we estimated perturbed away shapes. and compensated distortion by using estimated array shapes and synthesized arrays in aperture domain. As experimental data, we used the one obtained from towed array in neighboring waters of the Korean peninsula. We extended array by compensating differences in time and spatial position between overlapped subarrays by using SAS techniques. In simulation results. we confirmed that the bearing resolution was enhanced.
Audio Contents Adaptation Technology According to User′s Preference on Sound Fields
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 437~445
In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.
Adaptive Enhancement Algorithm of Perceptual Filter Using Variable Threshold
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 446~453
In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.
An Acoustic Analysis on the Korean 8 Monophthongs - With Respect to the Acoustic Variables on the F1/F2 Vowel Space -
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 454~461
Study on the Improvement of Speech Recognizer by Using Time Scale Modification
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 462~472
In this paper a method for compensating for thp performance degradation or automatic speech recognition (ASR) is proposed. which is mainly caused by speaking rate variation. Before the new method is proposed. quantitative analysis of the performance of an HMM-based ASR system according to speaking rate is first performed. From this analysis, significant performance degradation was often observed in the rapidly speaking speech signals. A quantitative measure is then introduced, which is able to represent speaking rate. Time scale modification (TSM) is employed to compensate the speaking rate difference between input speech signals and training speech signals. Finally, a method for compensating the performance degradation caused by speaking rate variation is proposed, in which TSM is selectively employed according to speaking rate. By the results from the ASR experiments devised for the 10-digits mobile phone number, it is confirmed that the error rate was reduced by 15.5% when the proposed method is applied to the high speaking rate speech signals.
PCMM-Based Feature Compensation Method Using Multiple Model to Cope with Time-Varying Noise
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 473~480
In this paper we propose an effective feature compensation scheme based on the speech model in order to achieve robust speech recognition. The proposed feature compensation method is based on parallel combined mixture model (PCMM). The previous PCMM works require a highly sophisticated procedure for estimation of the combined mixture model in order to reflect the time-varying noisy conditions at every utterance. The proposed schemes can cope with the time-varying background noise by employing the interpolation method of the multiple mixture models. We apply the‘data-driven’method to PCMM tot move reliable model combination and introduce a frame-synched version for estimation of environments posteriori. In order to reduce the computational complexity due to multiple models, we propose a technique for mixture sharing. The statistically similar Gaussian components are selected and the smoothed versions are generated for sharing. The performance is examined over Aurora 2.0 and speech corpus recorded while car-driving. The experimental results indicate that the proposed schemes are effective in realizing robust speech recognition and reducing the computational complexities under both simulated environments and real-life conditions.
Bandwidth Scalable Wideband Speech Codec
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 481~487
In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1
Error-Tolerant Music Information Retrieval Method Using Query-by-Humming
The Journal of the Acoustical Society of Korea, volume 23, issue 6, 2004, Pages 488~496
This paper describes a music information retrieval system which uses humming as the key for retrieval Humming is an easy way for the user to input a melody. However, there are several problems with humming that degrade the retrieval of information. One problem is a human factor. Sometimes people do not sing accurately, especially if they are inexperienced or unaccompanied. Another problem arises from signal processing. Therefore, a music information retrieval method should be sufficiently robust to surmount various humming errors and signal processing problems. A retrieval system has to extract pitch from the user's humming. However pitch extraction is not perfect. It often captures half or double pitches. even if the extraction algorithms take the continuity of the pitch into account. Considering these problems. we propose a system that takes multiple pitch candidates into account. In addition to the frequencies of the pitch candidates. the confidence measures obtained from their powers are taken into consideration as well. We also propose the use of an algorithm with three dimensions that is an extension of the conventional DP algorithm, so that multiple pitch candidates can be treated. Moreover in the proposed algorithm. DP paths are changed dynamically to take deltaPitches and IOIratios of input and reference notes into account in order to treat notes being split or unified. We carried out an evaluation experiment to compare the proposed system with a conventional system. From the experiment. the proposed method gave better retrieval performance than the conventional system.