Abstract: A system (1) is described which is suited for suppressing audio distortion. The system comprises echo cancelling means g1, g2 coupled between an audio output (4) and a distorted desired audio sensing microphone array (3), and a filter arrangement (7) coupled to the echo cancelling means g1, g2 and/or the microphone array (3). The filter arrangement (7) includes filter coefficients representing at least a part of the audio distortion, such as reverberation. The system also comprises an at least partly mirrored circuit arrangement g?, 7? for copying thereby simulated audio distortion representative filter coefficient values into the filter coefficients of said filter arrangement (7). Such copied values can be then used for suppressing reverberation in a distorted desired signal by the filter arrangement (7).

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: A device for and method of calibrating a microphone, comprising a loudspeaker (3) for converting a loudspeaker input signal (5) into sound; a microphone (4) for converting received sound into a microphone output signal (16), and calibration means for calibrating an output power of the microphone relative to a desired power level. The calibration means comprise impulse response estimating means (7) for estimating an acoustic impulse response of the microphone by correlating the microphone output signal (6) and the loudspeaker input signal (5) when the microphone (4) receives the sound from the loudspeaker (3), whereby the output power of the microphone (4) is estimated.

Abstract: An interference canceller is described, comprising an adaptive filter for modeling an interference, such as an echo or noise, and a spectral processor for processing the modeled interference together with near end speech and the interference. The interference canceller further comprises an interference model mismatch compensator coupled to the adaptive filter for providing a mismatch signal to the spectral processor, said mismatch signal showing a speech independent decay. With the model mismatch signal of the interference canceller the interference power spectrum can be estimated very accurately leading to a significant convergence improvement of the acoustic canceller. Especially in the initial convergence phase at the start of a communication session a high quality operation of the acoustic canceller is achieved, which is important as this determines the first quality impression of the user.