A system for and a method of generating an audio image for use in rendering audio. The method comprises accessing an audio stream; accessing positional information, the positional information comprising a first position, a second position and a third position; and generating an audio image. In some embodiments, generating the audio image comprises generating, based on the audio stream, a first virtual wave front to be perceived by a listener as emanating from the first position; generating, based on the audio stream, a second virtual wave front to be perceived by the listener as emanating from the second position; and generating, based on the audio stream, a third virtual wave front to be perceived by the listener as emanating from the third position.

Methods and apparatus to analyze microphone placement for watermarks and signatures are disclosed. An example instructions cause one or more processors to at least determine a variance of a magnitude spectrum of a frequency band corresponding to a frequency spectrum of a first audio signal sensed with an audio sensor. The example instructions further cause the one or more processors to determine, based on the variance, a recovery rate associated with at least one of watermark detection or signature generation to be performed on a second audio signal to be sensed with the audio sensor.

Systems, apparatuses and methods may provide away to render augmented reality and virtual reality (VR/AR) environment information. More particularly, systems, apparatuses and methods may provide a way to selectively suppress and enhance VR/AR renderings of n-dimensional environments. The systems, apparatuses and methods may deepen a user's VR/AR experience by focusing on particular feedback information, while suppressing other feedback information from the environment.

The invention provides a method for generating output filters to a plurality of loudspeakers at respective positions for playback of a plurality of different input signals in respective spatially different sound zones by means of a processor system. The method comprising computing spatio-temporal correlation matrices in response to spatial information, e.g. measured transfer functions, and in response to desired sound pressures in the plurality of sound zones. Joint eigenvalue decomposition of the spatial correlation matrices are then computed, or at least an approximation thereof, to arrive at eigenvectors accordingly. Next, variable span filters a reformed from a linear combination of the eigenvectors in response to a desired trade-off between acoustic contrast and acoustic errors in the sound zones. Finally, output filter for each of the plurality of loudspeakers, for each of the plurality of input signals, in accordance with the variable span filters. The method is applicable also for optimization in one zone, e.g. for room equalization.

Provided is a sound pickup loudspeaker apparatus which suppresses the influence of feedback produced when adding a speaker while also reducing a sense of unnaturalness in a sound image. The sound pickup loudspeaker apparatus collects a target sound emitted from a first seat in the vehicle, plays back the collected target sound to a listener seated in a second seat in the vehicle from a second sound amplifying device disposed in a direction different from the first seat from the perspective of the listener, and furthermore plays back the collected target sound to the listener from a first sound amplifying device disposed in the same direction as the first seat from the perspective of the listener. The apparatus multiplies the target sound by a first gain and outputs the target sound to the first sound amplifying device; delays the target sound by an amount of time obtained by adding a time for achieving a precedence effect to a delay time of the first sound amplifying device relative to the second sound amplifying device; and multiplies the delayed target sound by a second gain and outputs the target sound to the second sound amplifying device. A first gain adjustment unit adjusts the first gain to a low value and a second gain adjustment unit adjusts the second gain to a high value.

The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.

According to one embodiment, a method, computer system, and computer program product for modulating external sounds to reflect the acoustic effects of virtual objects in a mixed-reality environment is provided. The present invention may include creating a knowledge corpus, recording a sound effect occurring externally to a mixed-reality environment experienced by a user operating the mixed-reality device; identifying one or more objects within the mixed-reality environment, including at least one virtual object; modulating the sound effect based on the knowledge corpus to simulate one or more acoustic effects of the one or more objects within the MR environment; and playing the modulated sound effect to the user.

An acoustic signal processing device calculates a signal waveform that a microphone receives when at least one of a sound source and the microphone is moving. The acoustic signal processing device includes a coefficient calculation unit configured to model a steering coefficient g.sub.k,m representing how much an amplitude of a sound source signal emitted at an mth discrete time, where m is an integer between 1 and M and M is a length of the sound source signal, is transferred to an amplitude of a signal that the microphone receives at a kth discrete time, where k is an integer between 1 and K and K is a length of a recording signal, using N-order Fourier series expansion where N is an integer of 1 or more, and a recording signal calculation unit configured to calculate the signal waveform that the microphone receives using the modeled steering coefficient g.sub.k,m.

A sound collection apparatus includes a substantially spherical base on which at least a predetermined number of recess portions are provided in a surface thereof with a predetermined interval therebetween; and a predetermined number of microphones. The predetermined number is an integer of 2 or greater. The microphones are installed on an inner bottom surface side of the recess portions one by one. Intervals between sound collecting portions of adjacent microphones are substantially equal to each other.

A microphone adapter connects to and receives analog audio input from a microphone and an external music source, such as a smart phone. The microphone audio input signal and external music audio input signal are converted to digital signals and multiplexed into a digital wireless signal that may be transmitted directly to multiple speakers. The digital wireless signals are uncompressed, high definition digital audio signals transmitted in the 2.4 GHz band with a range of several hundred feet. The microphone adapter enables user to easily establish a pop-up public address system without the need for speaker cables, sound boards, mixers, and the like.