摘要:
The application relates to a method for audio signal processing. The application further relates to a method of processing signals obtained from a multi-microphone system. The object of the present application is to reduce undesired noise sources and residual echo signals from an initial echo cancellation step. The problem is solved by receiving M communication signals in frequency subbands where M is at least two; processing the M subband communication signals in each subband with a blocking matrix ( 203,303,403 ) of M rows and N linearly independent columns in each subband, where N >=1 and N M , to obtain N target-cancelled signals in each subband; processing the M subband communication signals and the N target-cancelled signals in each subband with a set of beamformer coefficients ( 204,304,404 ) to obtain a beamformer output signal in each subband; processing the communication signals with a target absence detector ( 309 ) to obtain a target absence signal in each subband; using the target absence signal to obtain an inverse target-cancelled covariance matrix of order N ( 310,410 ) in each band; processing the N target-cancelled signals in each subband with the inverse target-cancelled covariance matrix in a quadratic form ( 312, 412 ) to yield a real-valued noise correction factor in each subband; using the target absence signal to obtain an initial estimate ( 311, 411 ) of the noise power in the beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with the noise correction factor to obtain a refined estimate ( 417 ) of the power of the beamformer output noise signal component in each subband; processing the refined estimate of the power of the beamformer output noise signal component with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value ( 206,306,406 ) to obtain a postfilter output signal in each subband; processing the postfilter output subband signals through a synthesis filterbank ( 207,307,407 ) to obtain an enhanced beamformed output signal where the target signal is enhanced by attenuation of noise signal components. This has the advantage of providing improved sound quality and reduction of undesired signal components such as the late reverberant part of an acoustic echo signal. The invention may e.g. be used for headsets, hearing aids, active ear protection systems, mobile telephones, teleconferencing systems, karaoke systems, public address systems, mobile communication devices, hands-free communication devices, voice control systems, car audio systems, navigation systems, audio capture, video cameras, and video telephony.
摘要:
Disclosed is a method of reducing feedback in a hearing aid adapted to be worn by a user, the method comprising the step of: receiving an audio input signal in an input transducer in the hearing aid; wherein the method further comprises the steps of: transforming the input signal into the frequency domain; dividing the audio signal into a plurality of frequency bands; determining a threshold frequency over which a plurality of upper frequency bands lies; multiplying each of the plurality of upper frequency bands by a random phase, thereby obtaining a plurality of phase randomized upper frequency bands; synthesizing the plurality of phase randomized upper frequency bands and the lower frequency bands to an output signal; transforming the output signal into the time-domain; and transmitting the output signal to an output transducer of the hearing aid.
摘要:
The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for the speech processing, e.g. in a hearing instrument.
摘要:
Disclosed is a method of reducing feedback in a hearing aid system comprising left and right hearing aids, each hearing aid being adapted to be worn by a user and for communicating with each other, the method comprising the step of: receiving an audio input signal in an input transducer in the hearing aid; wherein the method further comprises the steps of: transforming the input signal into the frequency domain; dividing the audio signal into a plurality of frequency bands; determining a threshold frequency over which a plurality of upper frequency bands lies; multiplying each of the plurality of upper frequency bands by a random phase, thereby obtaining a plurality of phase randomized upper frequency bands; synthesizing the plurality of phase randomized upper frequency bands and the lower frequency bands to an output signal; transforming the output signal into the time-domain; and transmitting the output signal to an output transducer of the hearing aid, wherein the same random phase is changed by the same amount in the left and the right hearing aids for each upper frequency band. A hearing aid system comprising left and right hearing aids adapted to communicate with each other is further more disclosed.
摘要:
The invention relates to a method of identifying and correcting errors in a noisy binary mask. An object of the present invention is to provide a scheme for improving a binary mask representing speech. The problem is solved in that the method comprises a) providing a noisy binary mask comprising a binary representation of the power density of an acoustic signal comprising a target signal and a noise signal at a predefined number of discrete frequencies and a number of discrete time instances; b) providing a statistical model of a clean binary mask representing the target signal; and c) using the statistical model to detect and correct errors in the noisy binary mask. This has the advantage of providing an alternative and relatively simple way of improving an estimate of a binary mask representing a speech signal. The invention may e.g. be used for the speech processing, e.g. in a hearing instrument.
摘要:
Disclosed is method of generating an audible signal in a hearing aid by estimating a weighting function of received audio signals, the hearing aid is adapted to be worn by a user; the method comprises the steps of: estimating a directional signal by estimating a weighted sum of two or more microphone signals from two or more microphones, where a first microphone of the two or more microphones is a front microphone, and where a second microphone of the two or more microphones is a rear microphone; estimating a direction-dependent time-frequency gain, and synthesizing an output signal;
wherein estimating the direction-dependent time-frequency gain comprises: • obtaining at least two directional signals each containing a time-frequency representation of a target signal and a noise signal; and where a first of the directional signals is defined as a front aiming signal, and where a second of the directional signals is defined as a rear aiming signal; • using the time-frequency representation of the target signal and the noise signal to estimate a time-frequency mask; and • using the estimated time-frequency mask to estimate the direction-dependent time-frequency gain.