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.

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 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.

Various examples related to determining a location of an active participant are provided. In one example, image data of a room from an image capture device is received. First audio data from a first microphone array at the image capture device is received. Second audio data from a second microphone array spaced from the image capture device is received. Using a three dimensional model, a location of the second microphone array is determined. Using the first audio data, second audio data, location of the second microphone array, and an angular orientation of the second microphone array, an estimated location of the active participant is determined.

A system and method for joint acoustic echo control and adaptive array processing, comprising the decomposition of a captured sound field into N sub-sound fields, applying linear echo cancellation to each sub-sound field, selecting L sub-sound fields from the N sub-sound fields, performing L channel adaptive array processing utilizing the L selected sub-sound fields, and applying non-linear audio echo cancellation.

In accordance with embodiments of the present disclosure, a method may include determining capability of each particular information handling system of a plurality of information handling systems to support combination of functionality of audio components of the particular information handling system. The method may also include determining audio performance parameters of the audio components of the information handling systems of the plurality of information handling systems which are capable of supporting combination of functionality of audio components. The method may further include, based on the audio performance parameters, combining functionality of audio components of the plurality of information handling systems which are capable of supporting combination of functionality of audio components.

A data transmission method, where a host acquires parameter information of a wireless communication channel between a wireless microphone array and the host, that is, a signal-to-noise ratio or bandwidth. The host reduces sampling frequency of the wireless microphone array or decreases a quantity of data transmission paths between the wireless microphone array and the host when the acquired parameter information satisfies a first preset condition such that bandwidth occupied when the wireless microphone array transmits data is reduced.

Various examples related to determining a location of an active speaker are provided. In one example, image data of a room from an image capture device is received and a three dimensional model is generated. First audio data from a first microphone array at the image capture device is received. Second audio data from a second microphone array laterally spaced from the image capture device is received. Using the three dimensional model, a location of the second microphone array with respect to the image capture device is determined. Using the audio data and the location and angular orientation of the second microphone array, an estimated location of the active speaker is determined. Using the estimated location, a setting for the image capture device is determined and outputted to highlight the active speaker.

A conferencing system includes a display device that displays video received from a remote communication device of a communication partner. An audio stream is transmitted to the remote communication device. The audio stream includes real-world sounds produced by one or more real-world audio sources captured by a microphone array and virtual sounds produced by one or more virtual audio sources. A relative volume of sounds in the audio stream is selectively adjusted based, at least in part, on real-world positioning of corresponding audio sources, including real-world and/or virtualized audio sources.

Systems and methods are provided to determine a subset of D microphones in a set of N microphones on a perimeter of a space to monitor a target location. The space is divided into L interference locations. An equation is solved to determine microphone weights for the N microphones by minimizing the maximum gain for signals related to the target location and interference locations, further optimized over an l.sup.1 penalty by applying a Lagrange multiplier to an l.sup.1 norm of the microphone weights in a manner that determines a set of D non-zero microphones weights and a set of (N-D) microphone weights that are zero or close to zero. Microphone weights are determined for at least 2 different frequencies.