Abstract: This disclosure relates to measurement of maximum stable gain of a hearing assistance device, including but not limited to hearing aids, as a function of frequency. In various approaches an adaptive filter with a variable step size is used to determine maximum stable gain as a function of frequency. In various approaches, the determination is done in process steps performed by the hearing assistance device. In various approaches, the determination is done in process steps performed by the hearing assistance device and by a host computer.

Abstract: The present invention relates to a telephony device comprising a near-mouth microphone (M1) for picking up an input acoustic signal including the speaker's voice signal (S1) and an unwanted noise signal (N1,D1), a far-mouth microphone (M2) for picking up an unwanted noise signal (N2,D2) in addition to the near-end speaker's voice signal (S2), said speaker's voice signal being at a lower level than the near-mouth microphone, and an orientation sensor for measuring an orientation indication of said mobile device. The telephony device further comprises an audio processing unit comprising an adaptive beamformer (BF) coupled to the near-mouth and far-mouth microphones, including spatial filters for spatially filtering the input signals (z1,z2) delivered by the two microphones, and a spectral post-processor (SPP) for post-processing the signal delivered by the beam-former so as to separate the desired voice signal from the unwanted noise signal so as to deliver the output signal (y).

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: Disclosed is an echo canceller comprising two or more adaptive filters for calculating echo estimates, whereby the adaptive filters each have adaptation control mechanisms for applying individual update control criteria. The adaptive filters are arranged in series. Each of the adaptation control mechanisms of the adaptive filters may apply individual update control criteria for both direct echo and diffuse echo. Several step-size reduction strategies are presented.

Abstract: A two way sound reproduction system, such as a mobile phone system has an input (2) for an incoming far end signal (W), a loudspeaker (5), a D-to-A-converter (3) between the input (2) and the loudspeaker (5), a microphone (7), a A-to-D (8) converter after the microphone (7), an echo cancellation system (AEC) and an output (12) for an outgoing far end signal. The system comprises a pre-processor (30) between the input (2) and the D-to-A converter (3) comprising: an amplifier (32) to amplify the signal to a sufficient sound pressure level, a clipper or compressor or limiter (33) to limit the signal in the digital domain, so that the telephone system between D-to-A converter (3) and A-to-D converter (8) behaves substantially like a linear system.

Abstract: An echo canceller (1) is described, which has dedicated non stationary echo cancelling properties. The non stationary echo could be estimated directly, or indirectly by subtracting the stationary echo determined by a stationary noise estimator. The echo canceller may comprise an adaptive filter (4) and a residual echo processor (6) coupled to the adaptive filter, which residual echo processor is equipped with the non stationary echo canceller. Such a non stationary echo canceller is proposed in order to prevent the stationary component in the echo estimate, especially in the residual echo from continuously distorting near end speech. This improves echo canceller performance in terms of speech quality and speech intelligibility, which is particularly important in cases of single talk near end speech, as opposed to double talk.