This disclosure describes an invention that that mutes specific talkers using at least one beamforming microphone array 102 that is configured to generate N audio signals 108 where each audio signal is associated with a spatial pickup pattern 130, the microphone array(s) 102 are located in a room 200; a processor 104 and memory 105 operably coupled to the microphone array 102, the processor 104 configured to execute the following steps: (a) selectively mute or unmute an individual talker the room with a mute function 106 that controls whether to mute or unmute the individual talker T1-T7 that is picked up by one or more of the individual audio signals, the mute function 106 includes speech learning that learns to identify different talkers in real time to allow the mute function 106 to identify transitions from one talker to another talker in the room 200, (b) output an audio signal 110 based on the selective muting of the talkers T1-T7 in the room 200.

A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter.

The present document relates to audio conference systems. In particular, the present document relates to the mapping of soundfields within an audio conference system. A conference multiplexer (110, 175, 210, 400) configured to place a first input soundfield signal (402) originating from a first soundfield endpoint (120, 170) within a 2D or 3D conference scene (300) to be rendered to a listener (301) is described. The first input soundfield signal (402) is indicative of a soundfield captured by the first soundfield endpoint (120, 170). The conference multiplexer (110, 175, 210, 400) is configured to set up the conference scene (300) comprising a plurality of talker locations (321, 322, 332, 331) at different angles (323, 333) with respect to the listener (301); provide a first sector (325); wherein the first sector (325) has a first angular width (324); wherein the first angular width (324) is greater than zero; and transform the first input soundfield signal (402) into a first output soundfield signal (403), such that for the listener (301) the first output soundfield signal (403) appears to be emanating from one or more virtual talker locations (321, 322) within the first sector (325).

A method and system for improved audio quality in teleconferencing are provided. The method includes analyzing the audio signal of multiple input lines in a teleconferencing system to detect if any two input lines contain substantially the same audio signal with a delay shorter than that of a conventional echo caused by an input line's own audio feedback via a teleconferencing server. The method further includes selecting the input line with the higher amplitude audio signal or the earlier received audio signal when two input lines with substantially the same audio signal are detected.

An audio processing system includes a group of microphones associated with a dynamic network of microphones and a receiver. The receiver is configured to identify a first signal received by a microphone in the plurality of microphones that is designated as a primary microphone, and identify a subset of microphones included in the plurality of microphones, where each microphone in the subset is associated with a respective signal corresponding to the first signal. The receiver is further configured to calculate a weighting factor for each microphone included in the subset based on the first signal and the respective signal and opportunistically establish a connection with a microphone associated with the dynamic network of microphones that is not included in the plurality of microphones; and, based on a signal received from this microphone, adjust a weighting factor for at least one of the microphones in the subset.

A video conference endpoint includes a microphone array to detect ultrasound transmitted by a user device and that is encoded with a user identity. The endpoint determines a position of the user device relative to the microphone array based on the detected ultrasound and recovers the user identity from the detected ultrasound. The microphone array also detects audio in an audio frequency range perceptible to humans from an active talker, and determines a position of the active talker relative to the microphone array based on the detected audio. The endpoint determines whether the position of the active talker and the position of the user device are within a predetermined positional range of each other. If it is determined that the positions are both within the predetermined positional range, the endpoint assigns the user identity to the active talker.

A conferencing system includes a plurality of microphones and an audio processing system that performs blind source separation operations on audio signals to identify different audio sources. The system processes the separated audio sources to identify or classify the sources and generates an output stream including the source separated content.

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.