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".
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The original paper is in English. Non-English content has been machine-translated and may contain typographical errors or mistranslations. Copyrights notice
Este artigo descreve um conjunto aprimorado de microfones com formação de feixe complementar baseado no novo algoritmo de adaptação de ruído. A formação de feixe complementar é baseada em dois tipos de formadores de feixe projetados para obter padrões de diretividade complementares entre si. Neste sistema, durante uma pausa na fala alvo, dois padrões de diretividade dos formadores de feixe são adaptados às direções de chegada do ruído, de modo que os valores esperados de cada espectro de potência de ruído sejam minimizados na saída da matriz. Usando esta técnica, podemos perceber os nulos direcionais para cada ruído mesmo quando o número de fontes sonoras excede o de microfones. Para avaliar a eficácia, experimentos de aprimoramento de fala e experimentos de reconhecimento de fala são realizados com base em simulações de computador com uma matriz de dois elementos e três fontes sonoras sob diversas condições de ruído. Em comparação com o formador de feixe adaptativo convencional e o método de subtração espectral convencional em cascata com o formador de feixe adaptativo, é mostrado que (1) o arranjo proposto melhora a relação sinal-ruído (SNR) da fala degradada em mais de 6 dB quando o ruído interferente são dois alto-falantes com SNR de entrada abaixo de 0 dB, (2) a matriz proposta melhora o SNR em cerca de 2 dB quando o ruído interferente é ruído de bolha e (3) uma melhoria na taxa de reconhecimento de mais de 18% é obtido quando o ruído interferente é de dois alto-falantes ou dois sinais sobrepostos de alguns alto-falantes, sob a condição de que o SNR de entrada seja de 10 dB.
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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
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@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},}
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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 -