A storage unit stores a first transfer function representing a transfer characteristic of a sound from a sound source for each sound source direction, a sound source direction estimating unit calculates a conversion coefficient of an acoustic signal for each channel in a frequency domain and a spatial spectrum for each sound source direction on the basis of the first transfer function and estimates a sound source direction in which the spatial spectrum becomes a maximum as an estimated sound source direction, a transfer function estimating unit estimates a transfer function for the estimated sound source direction as a second transfer function by normalizing the conversion coefficients among channels, and a transfer function updating unit updates the first transfer function for the estimated sound source direction using the second transfer function.

A sound effect adjustment method, device, electronic device, and storage medium are provided. The sound effect adjustment method can be applied to an electronic device. The method includes: detecting whether at least one target application is running on a current electronic device, wherein the current electronic device is capable of outputting audio sounds when the target application runs; determining at least one reference electronic device corresponding to the current electronic device when running the target application; obtaining sound effect parameters configured by the reference electronic device when running the target application; and adjusting, based on the sound effect parameters, output sound effects of the current electronic device when running the target application.

A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Techniques for controlling zone group and zone group characteristics such as audio volume in a multi-zone system are disclosed. The multi-zone system includes a number of multimedia players, each preferably located in a zone. A controller may control the operations of all of the zone players remotely from any one of the zones. Two or more zone players may be dynamically grouped as a zone group for synchronized operations. According to one aspect of the techniques, a zone group configuration can be managed, updated, modified via an interactive user interface provided in a controlling device. The zone group configuration may be saved in one of zone players. According to another aspect of the techniques, the audio volume control of a zone group can be performed individually or synchronously as a group.

Some disclosed methods involve multi-band bass management. Some such examples may involve applying multiple high-pass and low-pass filter frequencies for the purpose of bass input management. Some disclosed methods treat at least some low-frequency signals as audio objects that can be panned. Some disclosed methods involve panning low and high frequencies separately. Following high-pass rendering, a power audit may determine a low-frequency deficit factor that is to be reproduced by subwoofers or other low-frequency-capable loudspeakers.

A mechanism is described for facilitating display-based audio splitting in media environments, according to one embodiment. A method of embodiments, as described herein, includes receiving, at a computing device, a request to split audio streams associated with audio channels in communication with the computing device serving as a host device, wherein the audio streams are generated by one or more applications; selecting a split mode to facilitate splitting of the audio streams; and splitting the audio streams based on the split mode such that the audio streams are continuously mapped to the audio channels to adjust for delays and fine tuning.

Systems and methods for using auditorily-induced vection (AIV) to enhance a person's attitude awareness are provided herein. In at least one embodiment, an auditory object is projected based on the orientation of the person or a vehicle and the projected auditory is provided to the person. By projecting the auditory object, the attitude of the person or the vehicle can be conveyed to the person to enhance the person's attitude awareness.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Systems, apparatuses, and methods related to audio input prioritization are described. Systems may include to one or more sound devices. Audio input may be played on a sound device based on one or more settings applied to the audio input. A setting can be determined based on feedback received by a host. In an example, a method for prioritizing audio input may include receiving a plurality of audio inputs each comprising signaling representative of different sounds, and each from a different application, identifying an attribute of each audio input of the plurality of audio inputs, prioritizing each audio input based on the identified attribute of each audio input, and sending each audio input to a sound device to be played based on the prioritization.

The information processing device includes an acquisition unit, a determination unit, and a signal processing unit. The acquisition unit acquires one or a plurality of other sound elements. The determination unit determines importance levels of the sound elements acquired by the acquisition unit. The signal processing unit changes a sound source position of at least one of a sound element of a content being reproduced or another sound element according to the importance levels of the sound elements determined by the determination unit. Consequently, even in a case where the sound element of the reproduced content and the another sound element are simultaneously generated, sound interference between the sound element of the reproduced content and the another sound element can be suppressed.