The original paper is in English. Non-English content has been machine-translated and may contain typographical errors or mistranslations. ex. Some numerals are expressed as "XNUMX".
Copyrights notice
The original paper is in English. Non-English content has been machine-translated and may contain typographical errors or mistranslations. Copyrights notice
Kertas kerja ini menerangkan tatasusunan mikrofon pembentuk pancaran pelengkap yang dipertingkatkan berdasarkan algoritma penyesuaian hingar baharu. Pembentukan pancaran pelengkap adalah berdasarkan dua jenis pembentuk pancaran yang direka bentuk untuk mendapatkan corak kearah pelengkap yang berkaitan antara satu sama lain. Dalam sistem ini, semasa jeda dalam pertuturan sasaran, dua corak kearah pembentuk pancaran disesuaikan dengan arah kedatangan hingar supaya nilai jangkaan setiap spektrum kuasa hingar diminimumkan dalam output tatasusunan. Menggunakan teknik ini, kita boleh merealisasikan nol arah untuk setiap bunyi walaupun bilangan sumber bunyi melebihi mikrofon. Untuk menilai keberkesanan, eksperimen peningkatan pertuturan dan eksperimen pengecaman pertuturan dilakukan berdasarkan simulasi komputer dengan tatasusunan dua elemen dan tiga sumber bunyi di bawah pelbagai keadaan hingar. Berbanding dengan pembentuk pancaran penyesuaian konvensional dan kaedah penolakan spektrum konvensional yang dilantunkan dengan pembentuk pancaran suai, ditunjukkan bahawa (1) tatasusunan yang dicadangkan meningkatkan nisbah isyarat-ke-bunyi (SNR) pertuturan terdegradasi lebih daripada 6 dB apabila hingar mengganggu ialah dua pembesar suara dengan input SNR di bawah 0 dB, (2) tatasusunan yang dicadangkan meningkatkan SNR sebanyak kira-kira 2 dB apabila hingar yang mengganggu ialah hingar gelembung, dan (3) peningkatan dalam kadar pengecaman lebih daripada 18% diperoleh apabila bunyi yang mengganggu ialah dua pembesar suara atau dua isyarat bertindih bagi sesetengah pembesar suara di bawah syarat input SNR ialah 10 dB.
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Salinan
Hiroshi SARUWATARI, Shoji KAJITA, Kazuya TAKEDA, Fumitada ITAKURA, "Speech Enhancement Using Nonlinear Microphone Array Based on Noise Adaptive Complementary Beamforming" in IEICE TRANSACTIONS on Fundamentals,
vol. E83-A, no. 5, pp. 866-876, May 2000, doi: .
Abstract: This paper describes an improved complementary beamforming microphone array based on the new noise adaptation algorithm. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, during a pause in the target speech, two directivity patterns of the beamformers are adapted to the noise directions of arrival so that the expectation values of each noise power spectrum are minimized in the array output. Using this technique, we can realize the directional nulls for each noise even when the number of sound sources exceeds that of microphones. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations with a two-element array and three sound sources under various noise conditions. In comparison with the conventional adaptive beamformer and the conventional spectral subtraction method cascaded with the adaptive beamformer, it is shown that (1) the proposed array improves the signal-to-noise ratio (SNR) of degraded speech by more than 6 dB when the interfering noise is two speakers with the input SNR of below 0 dB, (2) the proposed array improves the SNR by about 2 dB when the interfering noise is bubble noise, and (3) an improvement in the recognition rate of more than 18% is obtained when the interfering noise is two speakers or two overlapped signals of some speakers under the condition that the input SNR is 10 dB.
URL: https://global.ieice.org/en_transactions/fundamentals/10.1587/e83-a_5_866/_p
Salinan
@ARTICLE{e83-a_5_866,
author={Hiroshi SARUWATARI, Shoji KAJITA, Kazuya TAKEDA, Fumitada ITAKURA, },
journal={IEICE TRANSACTIONS on Fundamentals},
title={Speech Enhancement Using Nonlinear Microphone Array Based on Noise Adaptive Complementary Beamforming},
year={2000},
volume={E83-A},
number={5},
pages={866-876},
abstract={This paper describes an improved complementary beamforming microphone array based on the new noise adaptation algorithm. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, during a pause in the target speech, two directivity patterns of the beamformers are adapted to the noise directions of arrival so that the expectation values of each noise power spectrum are minimized in the array output. Using this technique, we can realize the directional nulls for each noise even when the number of sound sources exceeds that of microphones. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations with a two-element array and three sound sources under various noise conditions. In comparison with the conventional adaptive beamformer and the conventional spectral subtraction method cascaded with the adaptive beamformer, it is shown that (1) the proposed array improves the signal-to-noise ratio (SNR) of degraded speech by more than 6 dB when the interfering noise is two speakers with the input SNR of below 0 dB, (2) the proposed array improves the SNR by about 2 dB when the interfering noise is bubble noise, and (3) an improvement in the recognition rate of more than 18% is obtained when the interfering noise is two speakers or two overlapped signals of some speakers under the condition that the input SNR is 10 dB.},
keywords={},
doi={},
ISSN={},
month={May},}
Salinan
TY - JOUR
TI - Speech Enhancement Using Nonlinear Microphone Array Based on Noise Adaptive Complementary Beamforming
T2 - IEICE TRANSACTIONS on Fundamentals
SP - 866
EP - 876
AU - Hiroshi SARUWATARI
AU - Shoji KAJITA
AU - Kazuya TAKEDA
AU - Fumitada ITAKURA
PY - 2000
DO -
JO - IEICE TRANSACTIONS on Fundamentals
SN -
VL - E83-A
IS - 5
JA - IEICE TRANSACTIONS on Fundamentals
Y1 - May 2000
AB - This paper describes an improved complementary beamforming microphone array based on the new noise adaptation algorithm. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, during a pause in the target speech, two directivity patterns of the beamformers are adapted to the noise directions of arrival so that the expectation values of each noise power spectrum are minimized in the array output. Using this technique, we can realize the directional nulls for each noise even when the number of sound sources exceeds that of microphones. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations with a two-element array and three sound sources under various noise conditions. In comparison with the conventional adaptive beamformer and the conventional spectral subtraction method cascaded with the adaptive beamformer, it is shown that (1) the proposed array improves the signal-to-noise ratio (SNR) of degraded speech by more than 6 dB when the interfering noise is two speakers with the input SNR of below 0 dB, (2) the proposed array improves the SNR by about 2 dB when the interfering noise is bubble noise, and (3) an improvement in the recognition rate of more than 18% is obtained when the interfering noise is two speakers or two overlapped signals of some speakers under the condition that the input SNR is 10 dB.
ER -