Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.

A method of multi-talker separation using a 3-tuple coprime microphone array, including generating, by a subarray signal processing module, a respective subarray data set for each microphone subarray of the 3-tuple coprime microphone array based, at least in part, on an input acoustic signal comprising at least one speech signal. The input acoustic signal is captured by the 3-tuple coprime microphone array. The 3-tuple coprime microphone array includes three microphone subarrays. The method includes determining, by the subarray signal processing module, a point by point product of the three subarray data sets; and determining, by the subarray signal processing module, a cube root of the point by point product to yield an acoustic signal output data. The acoustic signal output data has an output amplitude and an output phase corresponding to an input amplitude and an input phase of a selected speech signal of the at least one speech signal.

An apparatus and method for real-time audio processing employs a gaze detection sensor to detect a direction of a user's gaze and output a gaze signal corresponding to the detected direction of the user's gaze. A digital signal processing unit responds to a plurality of signals corresponding to a plurality of sounds received at the apparatus, and the determined direction of gaze to identify a signal of interest from the plurality of signals using the gaze signal. The signal of interest is processed for output to the user. In embodiments, a microphone array provides the plurality of signals. An imaging sensor may work with either the microphone array or the gaze detection sensor to identify the signal of interest.

A method for operating a beamforming microphone array for use in a predetermined area is provided herein, the method comprising: receiving acoustic audio signals at each of a plurality of microphones, converting the same to an electrical mic audio signal, and outputting each of the plurality of electrical mic audio signals; generating a user location data signal by a wave sensor system, and outputting the user location data signal, wherein the user location data signal includes location information of one or more people within the predetermined area; receiving both the user location data signal and plurality of echo-corrected mic audio signals at an adaptive beamforming device; and adapting one or more beams by the adaptive beamforming device based on the user location data signal and plurality of mic audio signals wherein each of the one or more beams acquires sound from one or more specific locations in the predetermined area.

An array of speaker modules includes a first speaker module having at least one electroacoustic driver and a housing that together define a first acoustic cavity having a first cavity volume. The array also includes a second speaker module having at least one electroacoustic driver and a housing that together define a first enclosed volume that is greater than the first cavity volume by a first volume difference. The second speaker module includes a first internal enclosure having a first internal volume that is substantially equal to the first volume difference. The portion of the first enclosed volume that is outside the first internal volume defines a second acoustic cavity having a second cavity volume that is substantially equal to the first cavity volume. This configuration enables the performance of low frequency acoustic drivers in the speaker modules to be substantially matched without requiring complicated alternative approaches for matched performance.

Audio spatial effects are provided using a gimbal-mounted ultrasonic speaker array in which a vertical line of ultrasonic speakers are provided on a speaker mount and are angled to direct sound at respective different elevation angles. The speaker mount can be rotated by a gimbal. In this way, the azimuth angle of the linear array is established in response to a control signal from, e.g., a game console or video player, with elevational angle of the desired sound beam being established by selecting one or more of the speakers in the linear array with the appropriate elevation angle.

A method performing automatic gain control (AGC) using an accelerometer in a headset starts with an accelerometer-based voice activity detector (VADa) generating a VADa output based on (i) acoustic signals received from at least one microphone included in a pair of earbuds and (ii) data output by at least one accelerometer that is included in the pair of earbuds. The at least one accelerometer detects vibration of the user's vocal chords. The headset includes the pair of earbuds. An AGC controller then performs automatic gain control (AGC) on the acoustic signals from the at least one microphone based on the VADa output. Other embodiments are also described.

The disclosure includes a sound system comprising at least one Digital Signal Processor control module, acting on a signal to the high-frequency transducer and on a signal to a mid-frequency transducer so as to apply, in a common frequency range, at least one magnitude parameter as well as at least one phase parameter so as to produce a directivity along the same angular sector as a directivity produced by the orientable shutters.

A system and method for driving a loudspeaker array across directivities and frequencies to maintain timbre constancy in a listening area is described. In one embodiment, a frequency independent room constant describing the listening area is determined using the directivity index of a first beam pattern, the direct-to-reverberant ratio DR at the listener's location in the listening area, and an estimated reverberation time T.sub.60 for the listening area at a designated frequency. On the basis of this room constant, an offset may be generated for a second beam pattern. The offset describes the decibel difference between first and second beam patterns to achieve constant timbre and may be used to adjust the second beam pattern at multiple frequencies. Maintaining constant timbre improves audio quality regardless of the characteristics of the listening area and the beam patterns used to represent sound program content. Other embodiments are also described.

A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.