• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.63-67
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• 1999

Recently, a method of measuring impulse response is widely used for a room acoustic evaluation instead of measuring reverberation time by white noise excitation. Comparing with the traditional reverberation time measurement, this method has many advantages such as good repeatability and the ability to extract various room acoustic parameters at one measurement. In this study, the author developed a measuring system that can extract mono-aural room acoustic parameters from an impulse response measured with MLS (Maximum Length Sequence) signal excitation. These room acoustic parameters include reverberation times(EDT, RT), speech intelligibilities(C50, C80, D, U50, U80, AI) and sound strength(G). This paper introduces the configuration of the developed measuring system, test results and discussions for the measurements at several rooms.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.687-693
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• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2E
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• pp.8-13
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• 2000

In this paper, we present an adaptive formant tracking algorithm for speech using closed phase WRLS-VFF-VT method. The pitch synchronous closed phase methods is known to give more accurate estimates of the vocal tract parameters than the pitch asynchronous method. However the use of a pitch-synchronous closed phase analysis method has been limited due to difficulties associated with the task of accurately isolating the closed phase region in successive periods of speech. Therefore we have implemented the pitch synchronous closed phase WRLS-VFF-VT algorithm for speech analysis, especially for formant tracking. The proposed algorithm with the variable threshold(VT) can provide a superior performance in the boundary of phone and voiced/unvoiced sound. The proposed method is experimentally compared with the other method such as two channel CPC method by using synthetic waveform and real speech data. From the experimental results, we found that the block data processing techniques, such as the two-channel CPC, gave reasonable estimates of the formant/antiformant. However, the data windows used by these methods included the effects of the periodic excitation pulses, which affected the accuracy of the estimated formants. On the other hand the proposed WRLS-VFF-VT method, which eliminated the influence of the pulse excitation by using an input estimation as part of the algorithm, gave very accurate formant/bandwidth estimates and good spectral matching.

• International Journal of Naval Architecture and Ocean Engineering
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• v.4 no.2
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• pp.96-111
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• 2012

Hull Mounted Sonar (HMS) is a long range submerged vehicle's hull-mounted passive sonar system which detects low-frequency noise caused by machineries of enemy ships or submerged vehicles. The HMS needs a sound absorption /insulation multi-layer structure to shut out the self-noise from own machineries and to amplify signals from outside. Therefore, acoustic analysis of the multi-layer system should be performed when the HMS is designed. This paper simplified the HMS multi-layer system to be an infinite planar multi-layer model. Also, main excitations that influence the HMS were classified into mechanical, plane wave and turbulent flow excitation, and the investigations for each excitation were performed for various models. Stiffened multi-layer analysis for mechanical excitation and general multi-layer analysis for turbulent flow excitation were developed. The infinite planar multi-layer analysis was expected to be more useful for preliminary design stage of HMS system than the infinite cylindrical model because of short analysis time and easiness of parameter study.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• The Journal of the Acoustical Society of Korea
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• v.25 no.6
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• pp.282-290
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• 2006

This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.34-40
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• 2010

As internet spread widely, various service which use the internet have been provided. One of the service is a internet phone. Its usage is increasing by the advantage of cost. But it has a falling off in quality of speech. because it use packet switching method while existing telephone use circuit switching method. Although vocoder use PLC (Packet Loss Concealment) algorithm, it has a weakness of continuous packet loss. In this paper, we propose methods to improve a lowering in quality of speech under continuous loss of packet by using PLC algorithm used in advanced G.729 and G.711. The proposed methods are LP (Linear Prediction) parameter interpolation, excitation signal reconstruction and excitation signal gain reconstruction. As a result, the proposed method shows superior performance about 11%.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.9-16
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• 2001

This paper deals with the inverse reconstruction of sectional area in nonuniform ducts by using the acoustical measurement. There have been many theoretical and experimental studies on the duct area reconstruction. In this research, the method using the impulse response function and area reconstruction algorithm was employed because of its mathematical and experimental simplicity. Based on the study results on the drawback of conventional impulse excitation method, a new measurement method is proposed, that uses the random noise source and the discrete inverse Fourier transform. It is found that the reconstruction errors of the present method is smaller than the conventional method. A random error analysis is performed in order to investigate the causes of reconstruction error and to clarify the applicable data range for area reconstruction.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.38-45
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• 1995

In this paper a way to improve the performance of the long term prediction is proposed, which adopts the Multi-band Excitation (MBE) method in addition to the Code-Excited Linear Prediction (CELP) method at low bit rates below 4.8 kbps. In the proposed method, the multiband long term prediction is performed on the periodic components which still remain after the long term prediction of the conventional CELP method. At this point, the whole frequency region is divided into subbands whose size is equal to the spacing between the harmonics of the fundamental frequency, and the periodic multiband excitation signals. are represented as the sum of sine waves approximately as large as the spectrum of the excitation signals, so that the actual characteristics of the excitation signals can be better taken into account. To evaluate the performance of the proposed method, computer simulation is performed at 4.8 kbps. The 4.8 kbps DoD CELP and the 4.4 kbps IMBE were chosen as the reference vocoders for the speech quality measure. The result of the perceptual speech quality measure showed that the performance of the proposed method is better than that of the 4.8 kbps DoD CELP vocoder, and similar to that of the 4.4 kbps IMBE vocoder.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.392-400
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• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.