• The Journal of the Acoustical Society of Korea
• /
• v.20 no.4
• /
• pp.48-53
• /
• 2001

In this paper, selectively attentive learning method has been proposed to improve the learning speed of multilayer Perceptron based on the error backpropagation algorithm. Three attention criterions are introduced to effectively determine which set of input patterns is or which portion of network is attended to for effective learning. Such criterions are based on the mean square error function of the output layer and class-selective relevance of the hidden nodes. The acceleration of learning time is achieved by lowering the computational cost per iteration. Effectiveness of the proposed method is demonstrated in a speaker adaptation task of isolated word recognition system. The experimental results show that the proposed selective attention technique can reduce the learning time more than 60% in an average sense.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.7
• /
• pp.37-46
• /
• 2001

This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.4
• /
• pp.23-30
• /
• 1999

In this study, we propose a new optimization algorithm for a withdrawal weighted SAW transversal filter to satisfy given, specifications such as bandwidth, ripple, insertion loss, and sidelobe rejection level. An analysis tool for the withdrawal weighted filter has been produced by means of the delta function model, and has been applied to the design of a filter consisting of an uniform input IDT and a withdrawal weighted output IDT. This optimization algorithm consists of three routines, which eventually determines eight design parameters to satisfy the performance specifications. At the first step, the number of input and output IDT fingers and their geometrical size are determined by the insertion loss specification. At the next step, the bandwidth is controlled by the change of the IDT finger position. Finally, the sidelobe rejection level is modified through the add/skip technique of IDT fingers. The algorithm in this paper is distinct from conventional techniques in that it can simultaneously consider all the specifications such as bandwidth, ripple, sidelobe rejection level and insertion loss, and optimize the geometry of the withdrawal weighted SAW filter.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.1
• /
• pp.29-38
• /
• 2012

This paper presents a transient noise reduction system in a speech signal. The proposed transient noise reduction system utilizes a median filter to reduce the transient noise. Since the median filter can distort speech during the noise reduction, a long-term prediction (LTP) filter is adopted as a pre-processor to minimize speech distortion. The speech information preserved by the LTP filter is re-synthesized after reducing the noise. This paper verifies the weakness of a linear prediction (LP) filter and the superiority of the LTP filter for preserving the speech component in transient noise presence environment. Applying the proposed system, the signal-to-noise ratio (SNR) of output is improved by 8dB in both speech and noise presence region, and PESQ score is increased by 1 point comparing with noisy input.

• Journal of Satellite, Information and Communications
• /
• v.11 no.3
• /
• pp.80-85
• /
• 2016

In this paper, by comparatively analyzing horn type speaker and line array type speaker developed based on line sound source theory and point sound source theory, we research whether theory is adaptable or not in real. Academically, point sound source is attenuated as much as 6dB in accordance with double distance and line sound source is attenuated as much as 3dB in accordance with double distance. Line array speaker system developed based on line sound source is analyzed by theory of line sound source about occurring small sound pressure attenuation and it is propose of research that array composition of right speaker is selected in accordance with use purpose and environment. For this purpose, we analyze theory of point sound source and line sound source. we analyze parameter value by simulating designed horn type speaker and line array speaker based on theory.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.2
• /
• pp.40-49
• /
• 1995

Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer charactersitics between the output and the error signal of the adaptive canceler. In this paper, we derive the filtered-x adaptive noise cancellation algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.1
• /
• pp.65-72
• /
• 1995

An auditory experiments for the phonetic value of the last consonants when its signal is transmitted through the amplifier from the last to the first, shortly speaking, time reversed waveform, were done for the 14 Korean consonants. Then the last consonant becomes to the first consonant in the time reversed waveform. The listeners who heard the 14 reversed consonants have recorded the phonetic value being heard. We analyzed these results by the method of articulation and the position of articulation. By the results, the phonetic value of the last consonants /n/, /l/ and /m/ is the same as the first consonants. Last consonant /d/ is heard like first consonant /n/. Last consonant /ng/ is heard like first consonant /m/. Last consonants /k/ and /b/ don't have any particular phonetic values. These results were tested by the experiments and were analyzed by the principle of articulation.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.2
• /
• pp.113-122
• /
• 1995

This paper describes characteristics of a bone conduction telephone which was developed for conductive hearing impaired persons to call without additional devices and results of its performance test. Not only the hearing impaired but also normal hearing persons can use this telephone because we developed a bone conduction vibrator with which they can perceive speech signal using functions of air conductive hearing as well as bone conductive hearing. It also has tone control function compensating hearing losses for the hearing impaired originating from their hearing characteristics, and using this function together with received volume control it has received volume range of 20dB in loudness rating, which is similar effect as what a telephone set with built-in received amplifier has. From results of articulation and intelligibility tests for 19 hearing impaired persons, we can see that if their bone-conduction hearing loss is 61dB or less, they can understand words or sentences and response well with this telephone.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.3
• /
• pp.285-292
• /
• 2002

One of the problems that should be solved in Text-To-Speech (TTS) is discontinuities at unit-joining points. To cope with this problem, a smoothing method using a low-pass filter is employed in this paper, In the proposed soothing method, a filter coefficient that controls the amount of smoothing is determined according to contort information to be synthesized. This method efficiently reduces both discontinuities at unit-joining points and artifacts caused by undesired smoothing. The amount of smoothing is determined with discontinuities around unit-joins points in the current synthesized speech and discontinuities predicted from context. The discontinuity predictor is implemented by CART that has context feature variables. To evaluate the performance of the proposed method, a corpus-based concatenative TTS was used as a baseline system. More than 6075 of listeners realized that the quality of the synthesized speech through the proposed smoothing is superior to that of non-smoothing synthesized speech in both naturalness and intelligibility.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.3
• /
• pp.293-303
• /
• 2002

This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.