Novel tools and techniques are provided for implementing voice monitoring and tracking, and, more particularly, to methods, systems, and apparatuses for implementing voice monitoring and tracking of participants in group settings. In various embodiments, a computing system might receive, from at least one audio sensor among the one or more audio sensors disposed within the first space, voice signals corresponding to voices associated with individuals present within the first space. The computing system might analyze the received voice signals to identify one or more individuals who are present within the first space. The computing system might present, within a user interface of the user device associated with the user, information regarding the identified one or more individuals to assist the user in coordinating discussions among the individuals present within the first space.

A system and method can provide nose, such as wind noise, mitigation and/or microphone blending. Some methods may include sampling a sound signal from a plurality of microphones to generate a frame comprising a plurality of time-frequency tiles of the sound signal, each time-frequency tile including respective values of at least one feature from the plurality of microphones, comparing the respective values of the at least one feature to determine whether each time-frequency tile satisfies a similarity threshold, and flagging each time-frequency tile as noise if it fails to satisfy the similarity threshold, grouping the plurality of time-frequency tiles into sets of frequency-adjacent time-frequency tiles, and for each set of frequency-adjacent time-frequency tiles in the frame: counting a number of flagged time-frequency tiles, and attenuating all of the time-frequency tiles in the each set if the number exceeds a noise bin count threshold to thereby reduce noise in the sound signal.

An electronic device is provided. The electronic device comprises a speaker, a plurality of microphones, at least one processor operatively connected with the speaker and the plurality of microphones, and a memory operatively connected with the at least one processor, wherein the memory is configured to store instructions which, when executed, cause the at least one processor to perform speech audio processing or non-speech audio processing on audio signals received via the plurality of microphones, upon obtaining a non-speech audio signal based on the speech audio processing or the non-speech audio processing, identify a non-speech audio signal pattern corresponding to the non-speech audio signal, obtain a non-speech audio signal-based first command based on the identified non-speech audio signal pattern, and perform at least one action corresponding to the obtained non-speech audio signal-based first command.

Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

A multi-channel microphone receiver (MCR) for two or more wireless microphones (M.sub.1, . . . , M.sub.N) comprises a network interface and at least one mixer (MX) adapted for mixing audio signals (D.sub.1, . . . , D.sub.N) of the microphones. The mixer may be configured without any reconfiguration of the actual network being required. In addition to the single audio channels, the audio signal mixed according to the configuration may be output via a separate audio output channel (DO.sub.Mx, AO.sub.Mx), which may be analog or digital.

In order to detect a replay attack in a speaker recognition system, at least one feature is identified in a detected magnetic field. It is then determined whether the at least one identified feature of the detected magnetic field is indicative of playback of speech through a loudspeaker. If so, it is determined that a replay attack may have taken place.

A method includes generating a synthesized non-reference high-band channel based on a non-reference high-band excitation corresponding to a non-reference target channel. The method further includes estimating one or more spectral mapping parameters based on the synthesized non-reference high-band channel and a high-band portion of the non-reference target channel. The method also includes applying the one or more spectral mapping parameters to the synthesized non-reference high-band channel to generate a spectrally shaped synthesized non-reference high-band channel. The method further includes generating an encoded bitstream based on the one or more spectral mapping parameters and the spectrally shaped synthesized non-reference high-band channel.

Techniques for presence-detection devices to detect movement of a person in an environment by emitting ultrasonic signals using a loudspeaker that is concurrently outputting audible sound. To detect movement by the person, the devices characterize the change in the frequency, or the Doppler shift, of the reflections of the ultrasonic signals off the person caused by the movement of the person. However, when a loudspeaker plays audible sound while emitting the ultrasonic signal, audio signals generated by microphones of the devices include distortions caused by the loudspeaker. These distortions can be interpreted by the presence-detection devices as indicating movement of a person when there is no movement, or as indicating lack of movement when a user is moving. The techniques include processing audio signals to remove distortions to more accurately identify changes in the frequency of the reflections of the ultrasonic signals caused by the movement of the person.

An audio system includes an equatorial acoustic sensor array (EASA) that may be coupled to an object. The audio system is configured to detect, via the EASA, signals corresponding to a portion of a sound field in a local area. The detected signals are converted into a plurality of corresponding abstract representations that describe the portion of the sound field. Effects of scattering of the object are removed from the abstract representations to create adjusted abstract representations. A set of spherical harmonic (SH) coefficients is determined using the adjusted abstract representations. The set of SH coefficients describe an entirety of the sound field. And the set of SH coefficients and head related transfer functions of a user are used for binaural rendering of the reconstructed sound field to the user.

Disclosed herein is a mixed reality application to use a multi-channel audio input to identify a character and origin of a given sound, then present a visual representation of the given sound on a near eye display. The visual representation including a vector to the source of the sound. The visual representation further including graphical elements that describe various attributes of the given sound including the magnitude, directionality, source, and threat level. Where the source of the given sound is moving, the visual representation shifts to illustrate the movement.