Abstract: An echo cancellation device including a first adaptive filter for producing a first echo cancellation signal, a second adaptive filter for producing a second echo cancellation signal, and a post-processor for suppressing any remaining echo. The first adaptive filter and the second adaptive filter are designed for canceling a first part of the echo impulse response and a second part of the echo impulse response respectively. The device may be utilized in a mobile telephone.

Abstract: Method and apparatus for determination of gain margin of a hearing assistance device under test. In varying examples, the impulse response for multiple levels can be taken and used to arrive at a gain margin. The method and apparatus, in various examples, process critical portions of the resulting data for efficient processing and to increase accuracy of measurements. The method and apparatus performing a plurality of measurements to determine impulse responses and to derive gain margin as a function of frequency therefrom. The present subject matter includes principles which may are adapted for use within a hearing assistance device using a single white noise stimulus, according to one example. The principles set forth herein can be applied to occluding and non-occluding hearing device embodiments. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: An acoustic echo cancellation device (1) for canceling an echo in a microphone signal (z) in response to a far-end signal (x) comprises: an adaptive filter unit (10) arranged for filtering the far-end signal (x) so as to produce an echo cancellation signal (y), —a combination unit (12) arranged for combining the echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), and a post-processor unit (13) arranged for substantially removing any remaining echoes from the residual signal. In accordance with the invention, the device further comprises: —a near-end pitch estimation unit (18) arranged for estimating the pitch of any near-end signal (s) contained in the microphone signal (z) and for controlling the post-processor unit (13) in dependence of the estimated pitch such that harmonics of the estimated pitch may be substantially preserved.

Abstract: An apparatus for acoustic beamforming comprises a beamform processor (105) for generating a beamformed signal from two audio inputs. An update processor (107) updates the beamforming filter of the beamform processor (105) if an update criterion is met. An adaptive filter (111) filters the signal from one of the signals and the difference signal between the filtered signal and the signal from the other audio input (101) is generated. An adaptation processor (115) adapts the adaptive filter (111) to minimize the difference signal. A criterion processor (109) modifies the update criterion in response to the (possibly normalized) difference signal. Specifically, the update criterion may be relaxed to improve acquisition performance if the difference signal is indicative of a strong signal outside the beam of the beamform processor (105).

Abstract: Method and apparatus for determination of gain margin of a hearing assistance device under test. In varying examples, the impulse response for multiple levels can be taken and used to arrive at a gain margin. The method and apparatus, in various examples, process critical portions of the resulting data for efficient processing and to increase accuracy of measurements. The method and apparatus performing a plurality of measurements to determine impulse responses and to derive gain margin as a function of frequency therefrom. The present subject matter includes principles which may are adapted for use within a hearing assistance device using a single white noise stimulus, according to one example. The principles set forth herein can be applied to occluding and non-occluding hearing device embodiments. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Method and apparatus for determination of gain margin of a hearing assistance device under test. In varying examples, the impulse response for multiple levels can be taken and used to arrive at a gain margin. The method and apparatus, in various examples, process critical portions of the resulting data for efficient processing and to increase accuracy of measurements. The method and apparatus performing a plurality of measurements to determine impulse responses and to derive gain margin as a function of frequency therefrom. The present subject matter includes principles which may are adapted for use within a hearing assistance device using a single white noise stimulus, according to one example. The principles set forth herein can be applied to occluding and non-occluding hearing device embodiments. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: An acoustic echo cancellation device (1) for canceling an echo in a microphone signal (z) in response to a far-end signal (x) comprises: an adaptive filter unit (10) arranged for filtering the far-end signal (x) so as to produce an echo cancellation signal (y),—a combination unit (12) arranged for combining the echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), and a post-processor unit (13) arranged for substantially removing any remaining echoes from the residual signal. In accordance with the invention, the device further comprises:—a near-end pitch estimation unit (18) arranged for estimating the pitch of any near-end signal (s) contained in the microphone signal (z) and for controlling the post-processor unit (13) in dependence of the estimated pitch such that harmonics of the estimated pitch may be substantially preserved.

Abstract: An apparatus for acoustic beamforming comprises a beamform processor (105) for generating a beamformed signal from two audio inputs. An update processor (107) updates the beamforming filter of the beamform processor (105) if an update criterion is met. An adaptive filter (111) filters the signal from one of the signals and the difference signal between the filtered signal and the signal from the other audio input (101) is generated. An adaptation processor (115) adapts the adaptive filter (111) to minimize the difference signal. A criterion processor (109) modifies the update criterion in response to the (possibly normalized) difference signal. Specifically, the update criterion may be relaxed to improve acquisition performance if the difference signal is indicative of a strong signal outside the beam of the beamform processor (105).

Abstract: An echo cancellation device (1) comprises a first adaptive filter (13) for producing a first echo cancellation signal (y1), a second adaptive filter (15) for producing a second echo cancellation signal (y2), and a post-processor (18) for suppressing any remaining echo. The first adaptive filter (13) and the second adaptive filter (15) are designed for canceling a first (e.g. direct) part of the echo impulse response and a second (e.g. diffuse) part of the echo impulse response respectively. The device (1) may be utilized in a mobile telephone.

Abstract: Hearing aid for improving the hearing ability of the hard of hearing, comprising an array of microphones, the electrical output signals of which are fed to at least one transmission path belonging to an ear. Means are provided for deriving two array output signals from the output signals of the microphones, the array having two main sensitivity directions running at an angle with respect to one another and each of which is associated to an array output signal. Each array output signal is fed to its own transmission path belonging to one ear of a person who is hard of hearing.

Abstract: Hearing aid for improving the hearing ability of the hard of hearing, comprising an array of microphones, the electrical output signals of which are fed to at least one transmission path belonging to an ear. Means are provided for deriving two array output signals from the output signals of the microphones, the array having two main sensitivity directions running at an angle with respect to one another and each of which is associated to an array output signal. Each array output signal is fed to its own transmission path belonging to one ear of a person who is hard of hearing.

Abstract: Disclosed is a communication system having stations mutually coupled through a communication channel, wherein at least one of the stations is provided with echo cancelling (EC) means. The EC means comprise: adaptive EC means arranged for simulating linear echo effects in said station, a subtracter coupled to the adaptive EC means and having a subtracter output, and non linear EC means coupled to the subtracter output for simulating additional echo effects in the communication station. The non linear EC means are arranged as harmonic suppressing post processor means coupled to the subtracter output for effecting non linear echo cancellation based on outputted higher harmonics. It is thus presented a low cost algorithm for full duplex acoustic echo cancellation, wherein a frequency dependent non linearity attenuation deals with non linearly distorted echo components. The system is well suited for robust hands free communication by mobile phones, video phones, conferencing phones etc.