Abstract: An apparatus comprising: an input configured to receive at least two microphone signals associated with at least one acoustic source; an audio source determiner configured to determine from at least part of the at least two microphone signals at least one audio source based on the at least one acoustic source; an audio source direction determiner configured to determine at least one direction associated with the determined at least one audio source; a calibrator configured to calibrate at least one of the at least two microphone signals based on the at least one direction.

Abstract: It is inter alia disclosed to determine a peak energy level for an audio frame of a band limited audio signal; determine that the peak energy level is a maximum peak energy level by determining that the peak energy level exceeds a peak energy level determined for at least one neighboring audio frame by a first predetermined energy threshold value; classify a region of audio samples associated with the maximum peak energy level as an audio click; and suppress audio samples of the region of audio samples classified as an audio click by multiplying at least one audio sample of the audio frame with a sample wise suppressor gain function.

Abstract: The present invention concerns a system for a conference call, which includes: at least one portable audio device (MEM2-MEMn) arranged in an common acoustic space (AS) which device (MEM2-MEMn) is equipped with audio components (LS2-LSn, MIC2-MICn) for inputting and outputting an audible sound and at least one communication module (22); at least one base station device (MA) to which at least the said one portable audio device is interconnected and which base station device is connected to the communication network (CN) in order to perform the conference call from the said common acoustic space. At least part of the portable audio devices are personal mobile devices which audio components (MIC2-MICn) are arranged to pick the audible sound from the said common acoustic space.

Abstract: The invention relates to echo cancelling methods used in telephone systems, more specifically to echo cancelling methods based on two filters. The echo cancelling method according to the invention employs two adaptive filters, namely, a main filter (10) and a background filter (50). The purpose of the background filter (50) is to make sure that the main filter (10) model stays correct. The main and background filter coefficients are updated on different conditions, and the residual signals (L.sub.-- RES1, L.sub.-- RES2) produced by the filters are compared to each other. The set of filter coefficients used for generating the echo canceller's output signal (S.sub.-- OUT) is selected on the basis of said comparison of the residual signals. The selection of the set of filter coefficients is carried out either by copying the background filter coefficients to the main filter or by interchanging the background and main filter functions.

Abstract: A method for adaptation to an echo location in an echo canceller comprising an output port, an input port, an adjustable delay unit (43) compensating for the pure delay component in the echo path between the echo location and the output port of the echo canceller, a digital filter (40) calculating, from a signal outgoing to the echo path, an echo location estimate which is shorter than the maximum allowed duration of the echo path. The echo location is determined by calculating the echo path location at which there is a high correlation between the signal outgoing to the echo path and the returned echo. The correlation is calculated in a sliding window having the length of the digital filter.