In at least one embodiment, an audio system is provided. The audio system includes a plurality of loudspeaker, a plurality of microphones, and an audio controller. The plurality of loudspeakers transmits an audio signal in a listening environment. The plurality of microphones detects the audio signal in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of the audio signal as the audio signal is played back through a first loudspeaker of the plurality of loudspeakers and to determine a second PPL of the audio signal as the audio signal is sensed by a first microphone of the plurality of microphones. The at least one audio controller is further configured to map the first loudspeaker of the plurality of loudspeakers to the first microphone of the plurality of microphones based at least on the first PPL and the second PPL.

Provided are an audio processing method and apparatus, and a storage medium, which relate to the technical field of artificial intelligence and, in particular, to the speech technical field. The specific implementation solution is as follows. In response to receiving to-be-processed audio, a target sounding direction corresponding to the to-be-processed audio is determined; direction sense reconstruction is performed on the to-be-processed audio according to a direction sense reconstruction filter corresponding to the target sounding direction to obtain target audio; and the target audio is output.

There is provided a sound tracing method. The method includes: a setup processing step of setting ray information; a ray generation step of generating a sound ray; a traversal/intersection test step of generating hit triangle information; a propagation path validation (PPV) step of searching a sound path; a guide plane sort step of generating and sorting a guide plane; a Reverb Geometry Collect/Reverb Plane Sort (RGC/RPS) step of generating and sorting a reverb plane; and an impulse response calculation step of calculating an impulse response (IR) and storing the calculated IR in a valid path buffer.

M audio signals are obtained by processing an audio signal by M virtual speakers; M first HRTFs and M second HRTFs are obtained, where the M first HRTFs corresponding to a left ear position, and the M second HRTFs corresponding to a right ear position; high-band impulse responses of some of the M first HRTFs are modified to obtain modified first target HRTFs, and high-band impulse responses of some of the M second HRTFs are modified to obtain modified second target HRTFs; a first target audio signal corresponding to the left ear position is obtained based on the modified first target HRTFs and un-modified first HRTFs, and the M audio signals; and a second target audio signal corresponding to the right ear position is obtained based on the modified second HRTFs, un-modified second target HRTFs, and the M audio signals.

Methods and apparatus assist listeners in distinguishing between electronically generated binaural sound and physical environment sound while the listener wears a wearable electronic device that provides the binaural sound to the listener. The wearable electronic device generates a visual alert or audio alert when the electronically generated binaural sound occurs.

Methods and systems are provided for augmenting voice output of a virtual character in an augmented reality (AR) scene. The method includes examining, by a server, the AR scene, said AR scene includes a real-world space and the virtual character overlaid into the real-world space at a location, the real-world space includes a plurality of real-world objects present in the real-world space. The method includes processing, by the server, to identify an acoustics profile associated with the real-world space, said acoustics profile including reflective sound and absorbed sound associated with real-world objects proximate to the location of the virtual character. The method includes processing, by the server, the voice output by the virtual character while interacting in the AR scene; the processing is configured to augment the voice output based on the acoustics profile of the real-world space, the augmented voice output being audible by an AR user viewing the virtual character in the real-world space. In this way, when the voice output of the virtual character is augmented, the augmented voice output may sound more realistic to the AR user as if the virtual character is physically present in the same real-world space as the AR user.

An audio system is proposed, dynamically playing optimized audio signals based on user position. A sensor circuits dynamically senses a target space to generate field context information. First speaker and second speaker are arranged for audio playback. A host device recognizes a user from the field context information, determines the user position corresponding to the target space, and adaptively assigns the user position as a target listening spot. A sensor circuit contains a camera capturing an ambient image out of the target space. A recognizer circuit analyzes the ambient image to obtain from the target space, the location, size and acoustic attribute information of an ambient object, allowing the control circuit to accordingly perform an object-based compensation operation on the target listening spot to generate optimized first channel audio signal and second channel audio signal.

An example playback device is configured to (i) receive, via a network interface, data representing a command to play back audio content, where the audio content is a first type of audio content, (ii) during playback of the first type of audio content via an audio amplifier configured to drive a speaker, apply a first calibration and a second calibration to playback by the playback device, where the first calibration at least partially offsets one or more acoustic characteristics of an environment surrounding the playback device when applied to playback by the playback device, and where the second calibration corresponds to the first type of audio content, and (iii) during playback of a second type of audio content via the audio amplifier configured to drive the speaker, apply a third calibration to playback by the playback device, where the third calibration corresponds to the second type of audio content.

A system that suppresses a plurality of interferences of different types in a received audio signal. The system comprises one or more microphones and an audio controller. The one or more microphones are configured to detect the audio signal. The audio controller applies an interference estimation algorithm to the audio signal to generate an attenuation coefficient for each of the plurality of interferences of different types. The audio controller applies the attenuation coefficients to the audio signal to generate an interference-suppressed audio signal in which the plurality of interferences of different types is suppressed. The audio controller determines a time domain signal based on the interference-suppressed audio signal to provide to an end user.

A device may be configured to play one or more of a plurality of audio streams. The device may include a memory configured to store the plurality of audio streams, each of the audio streams representative of a soundfield. The device also may include one or more processors coupled to the memory, and configured to present a user interface to a user, obtain an indication from the user via the user interface representing a desired listening position, select, based on the indication, at least one audio stream of the plurality of audio streams, and output, for a display and in response to obtaining the indication representing the desired listening position, a graphical user interface element suggesting an alternative listening position.