摘要:
The present invention relates to a microphone system, comprising at least one directional microphone with a first directivity to obtain a first microphone signal, at least one directional microphone with a second directivity that is worse than the first directivity, in particular, an omni-directional microphone, to obtain a second microphone signal, a detection means configured to receive the first and the second microphone signals and to output a detection signal indicating whether non-acoustic perturbations, in particular, wind noise, in the first and/or the second microphone signals exceed a predetermined level and a signal processing means configured to output an output signal generated from the first microphone signal and/or the second microphone signal on the basis of the detection signal. The invention also relates to a method reducing non-acoustic perturbations in microphone signals comprising detecting an audio signal by at least one directional microphone of a first directivity to obtain a first microphone signal, detecting the audio signal by at least one directional microphone with a second directivity that is worse than the first directivity, in particular, an omni-directional microphone, to obtain a second microphone signal, determining whether non-acoustic perturbations, in particular, wind noise, that exceed a predetermined level are present in the first and/or the second microphone signal and outputting an output signal generated from the first microphone signal and/or the second microphone signal on the basis of the result of the determining whether non-acoustic perturbations that exceed a predetermined level are present in the first and/or the second microphone signal.
摘要:
The present invention relates to a method for estimating signal coherence, comprising detecting sound generated by a sound source, in particular, a speaker, by a first microphone to obtain a first microphone signal x 1 (n) and by a second microphone to obtain a second microphone signal x 2 (n), filtering the first microphone signal x 1 (n) by a first adaptive filtering means, in particular, a first Finite Impulse Response filter, to obtain a first filtered signal Y 1 (e jΩ µ,k), filtering the second microphone signal x 2 (n) by a second adaptive filtering means, in particular, a second Finite Impulse Response filter, to obtain a second filtered signal Y 2 (e jΩ µ, k ) and estimating the coherence of the first filtered signal Y 1 (e jΩ µ,k) and the second filtered signal Y 2 (e jΩ µ,k), wherein the first and the second microphone signals x 1 (n) and x 2 (n) are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the filtered first and second filtered signals Y 1 (e jΩ µ, k ) and Y 2 (e jΩ µ, k ).
摘要:
The present invention relates a method for localization of a speaker in a room in that at least one loudspeaker and at least one microphone array are located, comprising the steps of outputting sound by the at least one loudspeaker such that the sound is at least partly reflected by the speaker; detecting the sound output by the at least one loudspeaker and at least partly reflected by the speaker by the microphone array to obtain microphone signals for each of the microphones constituting the microphone array; and determining the speaker's direction towards and/or distance from the microphone array on the basis of the microphone signals.
摘要:
The present invention relates to a method for voice control of at least one vehicular element outside a vehicular cabin, comprising arranging at least one microphone, in particular, a directional microphone, in a lamp housing installed in the vehicle body outside the vehicular cabin, detecting a verbal utterance by means of the at least one microphone to obtain at least one microphone signal, processing the at least one microphone signal for speech recognition of the verbal utterance to obtain a recognition result and automatically controlling the at least one vehicular element on the basis of the recognition result. The invention also provides a lamp housing in the vehicle body outside the vehicular cabin comprising one or more microphones.
摘要:
The present invention relates to a method for dereverberation of a microphone signal, comprising the steps of dividing the microphone signal into frames or sub-bands providing at least one loudspeaker signal estimating the reverberation energy of at least some of the frames or sub-bands on the basis of the at least one loudspeaker signal and filtering the microphone signal on the basis of the estimated reverberation energy of the at least some of the frames or sub-bands. The invention also relates to a signal processing means, comprising at least one microphone configured to obtain a microphone signal, at least one loudspeaker configured to output a loudspeaker signal, a reverberation estimating means configured to estimate the reverberation energy of the reverberation portion in the microphone signal on the basis of the loudspeaker signal and a dereverberation filtering means configured to receive the microphone signal and to reduce a reverberation portion in the microphone signal on the basis of the estimated reverberation energy.
摘要:
In order to improve the reliability of voice processing in a vehicle environment it is proposed to detect a sound signal in the vehicle environment and to identify a voice command, which originates from a vehicle user outside the vehicle, in the detected sound signal, taking into account a position information relating to the position of the vehicle user in the vehicle environment. Information on the position of the vehicle user may be obtained from a keyless-go-system (30) or another monitoring device (20) of the motor vehicle, for example an optical imaging device of a parking-assistance system or a driver-assistance system.
摘要:
The present invention relates to a system for signal processing of an acoustic input signal, comprising at least one microphone, an echo compensation filtering configured to receive at least one microphone signal from the at least one microphone and comprising echo compensation filter coefficients determined on the basis of the at least one microphone signal and an equalization filtering means configured to equalize the acoustic input signal and comprising equalization filter coefficients determined on the basis of the echo compensation filter coefficients. The invention also relates to a method for enhancing the quality of a first acoustic input signal, comprising outputting a second acoustic input signal by at least one loudspeaker to generate a loudspeaker signal, generating at least one microphone signal on the basis of the loudspeaker signal, echo compensating the at least one microphone signal by adaptation of an echo compensation filtering means and equalizing the first acoustic input signal on the basis of the adapted echo compensation filtering means.
摘要:
Speech dialog system, comprising: - a speech recognition unit recognising a user's speech input, - a speech output unit outputting speech to the user in response to the user's speech input, - a speech output control unit controlling a play back mode of the speech output to the user, - a detector detecting information from at least one of the speech input unit and a sensor detecting information about the surrounding of the speech dialog system, wherein the speech output control unit adapts the play back mode of the speech output unit in dependence on the information received from the detector.