A noise suppression device includes: an adaptive filter unit that suppresses, using an adaptive filter, a noise component contained in a voice signal generated from a voice captured by a voice input unit to generate a corrected voice signal; a noise generation detection unit that detects timing of generation of the noise component in the voice signal; and a period suppression unit that suppresses the corrected voice signal during a predetermined period of time after the timing of the generation of the noise component.

A voice receiving device configured for accurate listening includes a microphone array, a camera, a capturing module, a determining module, a time module, a calculating module, and a de-noising module. The microphone array captures a first voice signal and a second voice signal and the camera captures mouth pictures of a user. The determining module determines whether the first voice signal is synchronized with the mouth pictures, and if so compares the first voice signal to a model preset voice signal of a user to determine a target voice signal. The time module obtains time delay difference between one voice reaching different microphones. The calculating module calculates a position of sound source of the target voice signal. According to the position of the sound source, the de-noising module de-noises by reference to the second voice signal. The disclosure further provides a voice receiving method.

An electronic device including a voice obtaining microphone configured to receive a voice signal including a noise at a first level; a noise obtaining microphone configured to receive a voice signal including a noise at a second level higher than the first level; and a controller configured to estimate the noise at the second level from the voice signal received by the noise obtaining microphone, remove the noise at the first level from the voice signal received by the voice obtaining microphone based on the estimated noise, and perform an operation corresponding to the voice signal having the noise at the first level eliminated therefrom.

The disclosure describes methods, clients, and electronic devices for processing audio signals. One method for processing audio signals comprises: receiving a first audio signal inputted from a first audio acquisition terminal and a second audio signal inputted from a second audio acquisition terminal, wherein the first audio acquisition terminal and the second audio acquisition terminal are located in different positions of a same location; determining a target audio signal and a reference audio signal from the first audio signal and the second audio signal; determining a filter coefficient corresponding to the target audio signal based on the reference audio signal; and eliminating, from the target audio signal, a crosstalk signal determined based on the filter coefficient and the reference audio signal. The effect that a speech path can output speech signals with less interference is achieved.

From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained. An estimated first signal is generated by subtracting a pseudo second signal which is estimated to be mixed in a first mixed signal in which a first signal and a second signal are mixed from the first mixed signal. The pseudo second signal is obtained by a first adaptive filter using a second mixed signal in which the first signal and the second signal are mixed in a different proportion from the first mixed signal. A coefficient update amount of the first adaptive filter is made smaller as compared with a case when the estimated first signal is smaller than the first mixed signal, in case the estimated first signal is larger than the first mixed signal.

A video system and method are presented that uses an output-receiving microphone mounted on the housing of the video system to receive a signal representative of the output of a speaker in a use-environment. This signal is compared with a second signal received from a spoken-word microphone that is mounted more remotely from the speaker. In some embodiments, the spoken-word microphone is positioned in a low-pass filter tune pipe and is combined with a separate spoken-word microphone in a tuned microphone array so as to filter out frequencies not associated with the human voice. The two signals are magnitude matched, and the first signal is subtracted from the second to generate an improved voice signal for a voice recognition system.

Various implementations include systems for processing inner microphone audio signals. In particular implementations, a system includes an external microphone configured to be acoustically coupled to an environment outside an ear canal of a user; an inner microphone configured to be acoustically coupled to an environment inside the ear canal of the user; and an adaptive noise cancelation system configured to process an internal signal captured by the inner microphone and generate a noise reduced internal signal, wherein the noise reduced internal signal is adaptively generated in response to an external signal captured by the external microphone.

Provided is a noise reduction apparatus, a noise reduction method, and a program for generating voice that is easy to hear while reducing sudden sound. The noise reduction apparatus includes a first microphone configured to collect mainly voice and output the voice as a first signal, a second microphone configured to collect mainly sound other than the voice and output the sound as a second signal, a sudden sound detecting unit configured to detect sudden sound in the first signal and the second signal, a sudden sound information storage unit configured to store the sudden sound as one or a plurality of reference signals, and an adder configured to remove the sudden sound from the first signal based on the reference signal stored in the sudden sound information storage unit.

A system configured to improve audio processing by performing dereverberation and noise reduction during a communication session. The system may apply a two-channel dereverberation algorithm by calculating coherence-to-diffuse ratio (CDR) values and calculating dereverberation (DER) gain values based on the CDR values. While the device calculates the DER gain values prior to performing acoustic echo cancellation (AEC) processing, the device applies the DER gain values after performing residual echo suppression (RES) processing in order to avoid excessive attenuation of the local speech. To improve output speech quality, the device does not apply the DER gain values for nonreverberant signals, when a signal-to-noise ratio (SNR) value is too low, and/or when far-end talk (e.g., remote speech) is present. Dereverberation processing is further improved by using frequency dependent parameters to calculate the DER gain values and by adjusting other gain values when the DER gain values are applied.

A noise suppressing apparatus calculates a phase difference on the basis of a first and second sound signal obtained by a microphone array; calculates a first sound arrival rate on the basis of a first phase difference area and the phase difference and a second sound arrival rate on the basis of a second phase difference area and the phase difference; calculates a dissimilarity that represents a level of difference between the first sound arrival rate and the second sound arrival rate; determines whether the pickup target sound is included in the first sound signal on the basis of the dissimilarity; and determines a suppression coefficient to be applied to the frequency spectrum of the first sound signal, on the basis of a result of the determination of whether the pickup target sound is included and on the basis of the phase difference.