• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2008.11a
• /
• pp.137-138
• /
• 2008

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2008.11a
• /
• pp.139-140
• /
• 2008

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2014.04a
• /
• pp.360-361
• /
• 2014

• The Journal of the Acoustical Society of Korea
• /
• v.16 no.4
• /
• pp.35-41
• /
• 1997

Speech database is an indispensable part of speech research. Speech database is necessary to use in speech research and development processes, and to evaluate performances of various speech-processing systems. To use speech database for common purpose, it is necessary to design utterance list that has all the possible phonetical events in minimal number of words, and is independent of tasks. To meet those restrictions this paper extracts PBW set from large text corpus. Speech database that was constructed using PBW set for utterance list and its properties are described in this paper.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.7
• /
• pp.480-486
• /
• 2012

In the design of electro acoustic transducer, power factor improvement circuit is more required rather than impedance matching if the driving power amplifier has little inner resistance. Many research results have been focused on the power matching circuit designing for transferring maximum power in the wideband. There are few results in the designing study on the power factor improvement for the wide band electro acoustic transducer. In this paper, we propose a new design method on the power factor improvement for the wide band electro acoustic transducer. The proposed method consists of two steps, the chebyschev matching method and the constrained optimization, respectively.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.7
• /
• pp.472-479
• /
• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.3
• /
• pp.27-34
• /
• 1999

This paper proposes an algorithm to design the non-linear array so as to form efficiently the one-shot beam with relatively less sensors for acoustic measurement. In this algorithm, according to the spatial sampling theory the part for high frequency(HF) band has equispaced sensor array and the sensor distances below the HF band are decided as a function of number of HF sensors. As the results of the simulations, the mean and variances of directivity index(DI) of non-linear array which has less sensors are similar to those of linear array. and the DI variation for beam steering angle is very small. And the beam width at -2dB point is 6.8°. Thus it is confirmed that the design algorithm for non-linear array which is proposed to have less sensors can be efficiently used in acoustic measurement.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.3
• /
• pp.14-27
• /
• 1995

This paper presents the digital music sound DSP by using ADPCM and Perceptual Transform Corder in MPEG to compress sound data and minimize the quantization noise for musical instrument. these method are utilized to develop algorithm of equal power density envelope. And these results are applied to examine the specific characteristics of musical instrument and determine the compression method. The design of new RISC DSP which generates 32 voices of musical instrument simultaneously and the coding of 200 musical instrument sound data in 1MByte memory shows that these algorithm is very useful to regenerate musical sound by using the minimum size of memory.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06e
• /
• pp.223-226
• /
• 1998

In the Input Buffer Switch using the intial stage FIFO memory structure, It has pointed the Throughput limitation to the percent of 58.6 due to HOL(Head of Line) blocking in the DBP(Dedicated Buffer with Pointer) method, During that time, To overcome these problems, The prior papers have proposed the complicated Arbitration algorithms and Non-FIFO memory structures. and These showed the improved Throughput. But, Now, To design high speed ATM Switch which need to the tens of Giga bit/s or the tens of Tera bit/s. It has more difficulty in proceeding the priority of majority and the complicated Cell Scheduling, because of the problem in operating the control speed of the ratio of N to scanning each port and scheduling the Cell. In this paper, To overcome these problems, We could show more the improved performance than the existing DBP Window policy to design high speed ATM Switch.

• The Journal of the Acoustical Society of Korea
• /
• v.9 no.4
• /
• pp.33-44
• /
• 1990

In this paper, it represents the design requirements to make apartment houses comfortable in acoustic environments and to keep a privacy between dwellers. The sound insulation performance between rooms were tested in the fields and the conditions to satisfy the design requirements were investigated. For that purpose, the sound insulation performance between living room and bed room were investigated and noise transmittance according to the difference of arrangement in rooms were tested. Several flush doors and light-weighted walls which have difference with each material and structure were tested to suggest the methods that improve the acoustic environments.