Patent ID: 6847723
Filing Date: 2005-01-25
Classification: G01C,G10L

Abstract:
1. A voice input apparatus, comprising: a sound collecting means having first and second microphones; a means for eliminating surrounding noise having an adaptive filter to simulate a first noise signal produced from said first microphone by means of a second noise signal produced from said second microphone and having a first computing means to compute a difference between the first and second noise signals; and a means for eliminating speaker sound eliminating the output component of a speaker from a voice signal after said surrounding noise is removed by said means for eliminating surrounding noise, said speaker receiving an audio signal from an audio device; wherein said adaptive filter is a digital filter and performs an adaptive equalization processing for a voice signal produced from said second microphone, and a filter processing means updates the filter coefficient of said adaptive filter so that the voice signal produced from said second microphone and a difference signal produced from said first computing means are input to, and the power of said difference signal is minimized by, a LMS algorithm, said means for eliminating surrounding noise further comprising a portion for storing filter coefficients to separately store values for a filter coefficient of said adaptive filter, a transmittal characteristic from said speaker to said first microphone and a transmittal characteristic from said speaker to said second microphone; and wherein said means for eliminating speaker sound comprises a digital filter, the filter coefficient of which is calculated using the data stored in said portion for storing filter coefficients, and an input to which is said audio signal.