In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as\r\npossible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in\r\ndifferent frequency subbands, has shown promising results in this regard but can suffer from problems in\r\nsituations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in\r\norder to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved\r\nalgorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides\r\neven less speech distortion, better performance in a multi-speaker environment and improved noise suppression\r\nwhen speech is absent compared with previous work.
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