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Volume 25 , Issue 4
Showing 1-13 articles out of 13 articles from the selected issue
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TUTORIALS
PAPERS
  • Masashi Unoki, Masakazu Furukawa, Keigo Sakata, Masato Akagi
    Volume 25 (2004) Issue 4 Pages 232-242
    Released: July 01, 2004
    JOURNALS FREE ACCESS
    A basic method for restoring the power envelope from a reverberant signal was proposed by Hirobayashi et al. This method is based on the concept of the modulation transfer function (MTF) and does not require that the impulse response of an environment be measured. However this basic method has the following problems: (i) how to precisely extract the power envelope from the observed signal; (ii) how to determine the parameters of the impulse response of the room acoustics; and (iii) a lack of consideration as to whether the MTF concept can be applied to a more realistic signal. This paper improves this basic method with regard to these problems in order to extend this method as a first step towards the development for speech applications. We have carried out 1,500 simulations for restoring the power envelope from reverberant signals in which the power envelopes are three types of sinusoidal, harmonics, and band-limited noise and the carriers are white noise, to evaluate our improved method with regard to (i) and (ii). We then have carried out the same simulations in which the carriers are two types of carrier of white noise or harmonics with regard to (iii). Our results have shown that the improved method can adequately restore the power envelope from a reverberant signal and will be able to be applied for speech envelope restoration.
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  • Masashi Unoki, Keigo Sakata, Masakazu Furukawa, Masato Akagi
    Volume 25 (2004) Issue 4 Pages 243-254
    Released: July 01, 2004
    JOURNALS FREE ACCESS
    We previously proposed an improved method for restoring the power envelope from a reverberant signal, based on the modulation transfer function (MTF) concept in order to resolve the problems of Hirobayashi’s method. In this paper, to apply our improved method to reverberant speech, we consider three issues related to speech applications: (i) how to apply the improved method to speech dereverberation based on co-modulation characteristics; (ii) whether the MTF concept can also be applied in the sub-band for reverberant signals; and (iii) whether power envelope inverse filtering should be done separately in each channel. We propose an extended filterbank model based on these considerations. We have carried out 15,000 simulations of the power envelope restoration for reverberant speech signals, and our results have shown that the proposed model can adequately restore the power envelopes in all channels from reverberant speech signals. We also found that the estimation of the reverberation time should be done separately in each channel to improve the restoration accuracy of the power envelope.
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  • Emi Toyoda, Shinichi Sakamoto, Hideki Tachibana
    Volume 25 (2004) Issue 4 Pages 255-266
    Released: July 01, 2004
    JOURNALS FREE ACCESS
    In this paper, the measurement of sound absorption coefficient of acoustical materials in a reverberation room is investigated by numerical analysis based on the ray-tracing method. As a result, it has been confirmed that the sound absorption coefficient of the same specimen can much differ by the differences of room shape and by the effect of the sound diffusers. In order to evaluate the degree of sound diffusivity in a reverberation room, the angular dependence of the sound power incident to the room boundary and to the surface of the specimen have been calculated. From the result of the numerical analysis, the relationship between the measurement of sound absorption coefficient of materials and the diffusivity of the sound field in a reverberation room has been examined.
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  • Panuthat Boonpramuk, Tetsuo Funada, Hideyuki Nomura
    Volume 25 (2004) Issue 4 Pages 267-275
    Released: July 01, 2004
    JOURNALS FREE ACCESS
    In the present study, we attempt to sequentially estimate clean speech from noisy speech in various environments with neither a priori training nor estimation of noise power, and to improve quality of speech in both subjective and objective evaluations, respectively. The time varying auto-regressive with unknown input (ARUI) model and a new type of adaptive filter (called the “adaptive filter with bias”), proposed in the present paper, are applied for enhancement of speech that has been degraded by white/colored additive interference. The ARUI model is used to reduce high-frequency noise components. The adaptive filter is designed in order to reduce low-frequency additive noise components. The Kalman filter is used to estimate the parameters of the ARUI model and the adaptive filter. We confirmed that the quality of the enhanced speech is improved by comparison via original noisy speech or spectral subtraction in both objective and subjective evaluations, respectively.
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TECHNICAL REPORT
  • Tatsuya Hirahara
    Volume 25 (2004) Issue 4 Pages 276-285
    Released: July 01, 2004
    JOURNALS FREE ACCESS
    The IEC coupler, dummy-head and actual-ear responses, harmonic distortion, impulse response decay, phase rotation and group delay, external sound radiation, sound attenuation and acoustic crosstalk characteristics of six models of headphones, the TDH39, DT48, HD250 Linear II, HD414 Classic, HDA 200 and SR-Lambda Professional, are measured and compared. The results indicate that coupler responses and dummy-head responses are not identical to actual-ear responses in most models, open-air type headphones have poor sound shielding ability, and the harmonic distortion and phase characteristics of all headphones tested are not problematic.
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