• Title/Summary/Keyword: ASK-1

Search Result 770, Processing Time 0.025 seconds

Comparative analysis of inter-Korean acoustic terminology and proposal for integration (남북한 음향학 전문용어 비교 분석 및 통합안 제시)

  • Jiwan Kim
    • The Journal of the Acoustical Society of Korea
    • /
    • v.42 no.4
    • /
    • pp.271-284
    • /
    • 2023
  • This study compared 431 acoustic terminology of South Korean industrial standards and North Korean national standards based on IEC 60050-801:1994 international standards. In addition, this study attempted to integrate acoustic terminology between the two Koreas. There were 139 (32.3 %) AA types with exactly the same form of terminology, 35 (8.1 %) Aa types with different spellings due to differences in linguistic norms, and 257 (59.6 %) AB types with completely different forms. Morphologically, there were more than twice as many different types of terminology as the same type. In the integration of acoustic terminology with different forms, 178 (61 %) North Korean terminology and 76 (26 %) South Korean terminology were adopted. we would like to overcome the limitations of this study through the following suggestions. First, the government should support academic exchanges between the two Koreas and encourage the establishment of common standards for acoustic terminology. Second, efforts should be made to share acoustic terminology data between the two Koreas and publish an integrated acoustic terminology dictionary. Third, South and North Korea should jointly launch a terminology committee to make efforts to revise international standards together.

Evaluation of floor impact sound and airborne sound insulation performance of cross laminated timber slabs and their toppings (구조용 직교 집성판 슬래브와 상부 토핑 조건에 따른 바닥충격음 및 공기전달음 평가)

  • Hyo-Jin Lee;Yeon-Su Ha;Sang-Joon Lee
    • The Journal of the Acoustical Society of Korea
    • /
    • v.42 no.6
    • /
    • pp.572-583
    • /
    • 2023
  • Demand for wood in construction is increasing worldwide. In Korea, technical reviews of high-rise Cross Laminated Timber (CLT) buildings are under way. In this paper, Floor Impact Sound Insulation Performance (FISIP) and Transmission Loss (TL) of 150 mm thick CLT floor panels made of two domestic species, Larix kaempferi and Pinus densiflora, are investigated. The CLT slabs were tested in reverberation chambers connected vertically. When comparing Single Number Quantity (SNQ) of FISIP of the bare panels, the Larix CLT is 3 dB lower in heavy-weight and 1 dB in light-weight than the Pinus CLT. However, there was no difference when concrete toppings were added to improve the performance. As the concrete toppings became thicker, the heavy-weight was reduced by 9 dB ~ 20 dB, and the light-weight by 20 dB ~ 30 dB. And the analysis of these results with area density has confirmed that the area densities are highly correlated (R2 = 0.94 ~ 0.99) to the FISIP of the CLT. The types of CLT didn't affect the TL. Comparison of theoretical TL values with measured TL values has shown that the frequency characteristics are similar but 8 dB ~ 12 dB lower in measured values. The relationship between the TL and frequency characteristics of the tested CLT slabs was derived by using the correction value.

Highband Coding Method Using Matching Pusuit Estimation and CELP Coding for Wideband Speech Coder (광대역 음성부호화기를 위한 매칭퍼슈잇 알고리즘과 CELP 방법을 이용한 고대역 부호화 방법)

  • Jeong Gyu-Hyeok;Ahn Yeong-Uk;Kim Jong-Hark;Shin Jae-Hyun;Seo Sang-Won;Hwang In-Kwan;Lee In-Sung
    • The Journal of the Acoustical Society of Korea
    • /
    • v.25 no.1
    • /
    • pp.21-29
    • /
    • 2006
  • In this Paper a split bandwidth wideband speech coder and its highband coding method are Proposed. The coder uses a split-band approach. where the wideband input speech signal is split into two equal frequency bands from 0-4kHz and 4-8kHz. The lowband and the highband are coded respectively by the 11.8kb/s G.729 Annex E and the proposed coding method. After the LPC analysis, the highband is divided by two modes according to the properties of signals. In stationary mode. the highband signals are compressed by the mixture excitation model; CELP algorithm and W (Matching Pursuit) algorithm. The others are coded by the only CELP algorithm. We compare the performance of the new wideband speech coder with that of G.722 48kbps SB-ADPCM and G.722.2 12.85kbps in a subjective method. The simulation results show that the Performance of the proposed wideband speech coder has better than that of 48kbps G.722 and no better than that of 12.85kbps G.722.2.

Geoacoustic Inversion and Source Localization with an L-Shaped Receiver Array (L-자형 선배열을 이용한 지음향학적 인자 역산 및 음원 위치 추정)

  • Kim, Kyung-Seop;Lee, Keun-Hwa;Kim, Seong-Il;Kim, Young-Gyu;Seong, Woo-Jae
    • The Journal of the Acoustical Society of Korea
    • /
    • v.25 no.7
    • /
    • pp.346-355
    • /
    • 2006
  • Acoustic data from a shallow water experiment in the East Sea of Korea (MAPLE IV) is Processed to investigate the Performance of matched-field geo-acoustic inversion and source localization. The receiver array consists of two legs as in an L-shape. one vertical and the other horizontal lying on the seabed. Narrowband multi-tone CW source was towed along a slightly inclined bathymetry track. The matched-field geo-acoustic inversion includes comparisons between three processing techniques. all based on the Bartlett processor as; (1) the coherent processing of the data from the full array, (2) the incoherent Product of each output from both the horizontal and vertical arrays, and (3) the cross correlation between the horizontal and vertical arrays. as well as processing each array leg separately. To verify the inversion results. matched-field source localization for low level source signal components were performed using the same Processors used at the inversion stage.

Robust Speech Recognition Algorithm of Voice Activated Powered Wheelchair for Severely Disabled Person (중증 장애우용 음성구동 휠체어를 위한 강인한 음성인식 알고리즘)

  • Suk, Soo-Young;Chung, Hyun-Yeol
    • The Journal of the Acoustical Society of Korea
    • /
    • v.26 no.6
    • /
    • pp.250-258
    • /
    • 2007
  • Current speech recognition technology s achieved high performance with the development of hardware devices, however it is insufficient for some applications where high reliability is required, such as voice control of powered wheelchairs for disabled persons. For the system which aims to operate powered wheelchairs safely by voice in real environment, we need to consider that non-voice commands such as user s coughing, breathing, and spark-like mechanical noise should be rejected and the wheelchair system need to recognize the speech commands affected by disability, which contains specific pronunciation speed and frequency. In this paper, we propose non-voice rejection method to perform voice/non-voice classification using both YIN based fundamental frequency(F0) extraction and reliability in preprocessing. We adopted a multi-template dictionary and acoustic modeling based speaker adaptation to cope with the pronunciation variation of inarticulately uttered speech. From the recognition tests conducted with the data collected in real environment, proposed YIN based fundamental extraction showed recall-precision rate of 95.1% better than that of 62% by cepstrum based method. Recognition test by a new system applied with multi-template dictionary and MAP adaptation also showed much higher accuracy of 99.5% than that of 78.6% by baseline system.

Implementation of Parallel Processor for Sound Synthesis of Guitar (기타의 음 합성을 위한 병렬 프로세서 구현)

  • Choi, Ji-Won;Kim, Yong-Min;Cho, Sang-Jin;Kim, Jong-Myon;Chong, Ui-Pil
    • The Journal of the Acoustical Society of Korea
    • /
    • v.29 no.3
    • /
    • pp.191-199
    • /
    • 2010
  • Physical modeling is a synthesis method of high quality sound which is similar to real sound for musical instruments. However, since physical modeling requires a lot of parameters to synthesize sound of a musical instrument, it prevents real-time processing for the musical instrument which supports a large number of sounds simultaneously. To solve this problem, this paper proposes a single instruction multiple data (SIMD) parallel processor that supports real-time processing of sound synthesis of guitar, a representative plucked string musical instrument. To control six strings of guitar, we used a SIMD parallel processor which consists of six processing elements (PEs). Each PE supports modeling of the corresponding string. The proposed SIMD processor can generate synthesized sounds of six strings simultaneously when a parallel synthesis algorithm receives excitation signals and parameters of each string as an input. Experimental results using a sampling rate 44.1 kHz and 16 bits quantization indicate that synthesis sounds using the proposed parallel processor were very similar to original sound. In addition, the proposed parallel processor outperforms commercial TI's TMS320C6416 in terms of execution time (8.9x better) and energy efficiency (39.8x better).

The Performance Improvement of PLC by Using RTP Extension Header Data for Consecutive Frame Loss Condition in CELP Type Vocoder (CELP Type Vocoder에서 RTP 확장 헤더 데이터를 이용한 연속적인 프레임 손실에 대한 PLC 성능개선)

  • Hong, Seong-Hoon;Bae, Myung-Jin
    • The Journal of the Acoustical Society of Korea
    • /
    • v.29 no.1
    • /
    • pp.48-55
    • /
    • 2010
  • It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.

Mid Frequency Band Reverberation Model Development Using Ray Theory and Comparison with Experimental Data (음선 기반 중주파수 대역 잔향음 모델 개발 및 실측 데이터 비교)

  • Chu, Young-Min;Seong, Woo-Jae;Yang, In-Sik;Oh, Won-Tchon
    • The Journal of the Acoustical Society of Korea
    • /
    • v.28 no.8
    • /
    • pp.740-754
    • /
    • 2009
  • Sound in the ocean is scattered by inhomogeneities of many different kinds, such as the sea surface, the sea bottom, or the randomly distributed bubble layer and school of fish. The total sum of the scattered signals from these scatterers is called reverberation. In order to simulate the reverberation signal precisely, combination of a propagation model with proper scattering models, corresponding to each scattering mechanism, is required. In this article, we develop a reverberation model based on the ray theory easily combined with the existing scattering models. Developed reverberation model uses (1) Chapman-Harris empirical formula and APL-UW model/SSA model for the sea surface scattering. For the sea bottom scattering, it uses (2) Lambert's law and APL-UW model/SSA model. To verify our developed reverberation model, we compare our results with those in Ellis' article and 2006 reverberation workshop. This verified reverberation model SNURM is used to simulate reverberation signal for the neighboring seas of South Korea at mid frequency and the results from model are compared with experimental data in time domain. Through comparison between experiment data and model results, the features of reverberation signal dependent on environment of each sea is investigated and this analysis leads us to select an appropriate scattering function for each area of interest.

Speech Activity Decision with Lip Movement Image Signals (입술움직임 영상신호를 고려한 음성존재 검출)

  • Park, Jun;Lee, Young-Jik;Kim, Eung-Kyeu;Lee, Soo-Jong
    • The Journal of the Acoustical Society of Korea
    • /
    • v.26 no.1
    • /
    • pp.25-31
    • /
    • 2007
  • This paper describes an attempt to prevent the external acoustic noise from being misrecognized as the speech recognition target. For this, in the speech activity detection process for the speech recognition, it confirmed besides the acoustic energy to the lip movement image signal of a speaker. First of all, the successive images are obtained through the image camera for PC. The lip movement whether or not is discriminated. And the lip movement image signal data is stored in the shared memory and shares with the recognition process. In the meantime, in the speech activity detection Process which is the preprocess phase of the speech recognition. by conforming data stored in the shared memory the acoustic energy whether or not by the speech of a speaker is verified. The speech recognition processor and the image processor were connected and was experimented successfully. Then, it confirmed to be normal progression to the output of the speech recognition result if faced the image camera and spoke. On the other hand. it confirmed not to output of the speech recognition result if did not face the image camera and spoke. That is, if the lip movement image is not identified although the acoustic energy is inputted. it regards as the acoustic noise.

Modeling of Scattered Signal from Ship Wake and Experimental Verification (항적 산란신호의 모델링과 실험적 검증)

  • Ji, Yoon-Hee;Lee, Jae-Hoon;Kim, Jea-Soo;Kim, Jung-Hae;Kim, Woo-Shik;Choi, Sang-Moon
    • The Journal of the Acoustical Society of Korea
    • /
    • v.28 no.1
    • /
    • pp.10-18
    • /
    • 2009
  • A moving surface vessel generates a ship wake which contains a cloud of micro-bubbles with radii ranging between $8{\sim}200{\mu}m$. Such micro-bubbles can be detected by active sonar system for more than ten minutes depending on the size and speed of the surface vessel. In this paper, a reverberation model for the ship wake is presented. The developed model consists of the acoustic scattering model due to the distribution of the micro-bubbles and the kinematic model for the moving active sonar. The acoustic scattering model is based on the volume integration, where the volume scattering strengths are obtained from the spatial distribution of micro-bubbles. Since the directivity and look-direction of active sonar are important factors for moving active sonar, the kinematic model utilizes the Euler transformation to obtain the relative motion between the global and local coordinates. In order to verify the developed model, a series of sea experiment was executed in September 2007 to obtain the spatial-temporal distribution of a bubble cloud, and analyzed to be compared with the simulation results.