Acoustical Science and Technology
Online ISSN : 1347-5177
Print ISSN : 1346-3969
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Volume 29 , Issue 6
Showing 1-12 articles out of 12 articles from the selected issue
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PAPERS
  • Seigo Hirayama, Hitoshi Morinaga, Tadahiro Ohmi, Jun-ichiro Soejima
    Volume 29 (2008) Issue 6 Pages 345-350
    Released: November 15, 2008
    JOURNALS FREE ACCESS
    We measured the amount of hydroxyl radicals and sonoluminescence to investigate the condition of ultrasonic cavitation. The generation of OH· radicals and sonoluminescence upon ultrasonic radiation occurred with the same trend at various concentrations of N2 and O2 in water, so we found that sonoluminescence can be used to measure the amount of cavities as well as radicals. In the investigation of the cavity-generating area, in degassed water, cavities were generated at water surface, and in gas-dissolved water, cavities were weakly generated from the entire volume of water. As the result of varying the N2 concentration or the dissolving condition at the water surface, we found that cavities were generated maximally at a certain gas concentration, and then decreased with excess dissolved gases. Cavitation did not occur in the degassed water, but occurred when gases dissolved from the water surface.
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  • Xugang Lu, Masashi Unoki, Masato Akagi
    Volume 29 (2008) Issue 6 Pages 351-361
    Released: November 15, 2008
    JOURNALS FREE ACCESS
    To reduce speech degradation in reverberant environments, we previously proposed a modulation-transfer-function (MTF)-based method of speech dereverberation. By considering the temporal modulation properties of speech, and the exponential decay properties of the power envelope of the impulse response of room acoustics, we obtained the following MTF relation: the sub-band power envelope of reverberant speech that can be represented as a convolution between the sub-band power envelope of clean speech and the power envelope of the impulse response of room acoustics. On the basis of the MTF relation, inverse MTF filtering can be applied to restoring the power envelopes of reverberant speech. Therefore, the impulse response of the room acoustics in this restoration dose not need to be measured at any time since we model the power envelope of the impulse response as an exponential decay function. We have tested how effective this method is as a front-end for automatic speech recognition (ASR) systems in artificial and real reverberant environments. Reverberant speech signals were created by simply convoluting clean speech (AURORA-2J database) with the artificially produced or real impulse responses of room acoustics. A method based on the auditory power spectrum was used as a baseline for comparison. Compared with the baseline, the proposed method for artificial reverberant environments produced a 35.67% relative improvement in the error reduction rate (on average, for reverberation times from 0.2 to 2.0 s), and for real reverberant environments (43 reverberant impulse responses), it produced a 25.78% relative improvement in the error reduction rate. The results demonstrate that our new approach can improve the robustness of speech-recognition systems in reverberant environments, and it performs better than conventional methods.
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  • Yotaro Kubo, Masaaki Honda, Katsuhiko Shirai, Tomoyasu Komori, Nobumas ...
    Volume 29 (2008) Issue 6 Pages 362-371
    Released: November 15, 2008
    JOURNALS FREE ACCESS
    We propose three methods for implementing an improved MPEG-2/4 Advanced Audio Coding (AAC) encoder. The naive implementation using only the informative part of the AAC standard results in coded audio signals with a significantly poor quality. However, the encoder implementation can be modified to improve coding efficiency. Some coding modules require more sophisticated methods for improving the quality. The first among the three methods is a grouping method in the short block mode of AAC. The second is a codebook selection method in the Huffman coding. The third is the scale-factor sectioning method in bit allocation to the frequency components. We also present the methods proposed in our previous papers. The efficiency of the proposed methods has been evaluated objectively and subjectively. We have confirmed that the proposed methods are capable of improving the quality of encoded streams.
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TECHNICAL REPORT
  • Shugang Wei, Wenhai Xu
    Volume 29 (2008) Issue 6 Pages 372-377
    Released: November 15, 2008
    JOURNALS FREE ACCESS
    In this paper, we propose an FPGA design of an audio signal level compressor on the basis of an approximation algorithm using a polynomial. To implement a compression characteristic in a digital audio system, the gain calculation with fractional numbers is performed using a polynomial expression to approximate the power operation; then the compressor can be designed with a number of additions, multiplications and a division. The arithmetic circuits were implemented in FPGA technology, and the 16-bit compressor used 541 logic elements and 352 flip-flops of the hardware resource of EP20K200EFC484-2X from Altera. The proposed compressor can be applied as a functional unit in an on-chip audio system. The performance of the proposed compressor is evaluated by analysis and simulation.
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