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.

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 method of recording ambisonic sound fields with a spatially distributed plurality of ambisonic microphones comprising a step of recording sound signals from plurality of ambisonic microphones a step of converting recorded sound signals to ambisonic sound fields and a step of interpolation of the ambisonic sound fields according to the invention comprises a step of generating synchronizing signals for particular ambisonic microphones for synchronized recording of sound signals from plurality of ambisonic microphones and during the step of interpolation of the ambisonic sound fields it includes filtering sound signals from particular microphones with individual filters having a distance-dependent impulse response having a cut-off frequency f.sub.c(d.sub.m) depending on distance d.sub.m between point of interpolation and m-th microphone applying gradual distance dependent attenuation applying re-balancing with amplification of 0.sup.th ordered ambisonic component and attenuating remaining ambisonic components. Invention further concerns recording system and computer program product.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

A method, computer program product, and computing system for receiving information associated with an acoustic environment. Acoustic metadata associated with audio encounter information received by a first microphone system may be received. One or more speaker representations may be defined based upon, at least in part, the acoustic metadata associated with the audio encounter information and the information associated with the acoustic environment. One or more portions of the audio encounter information may be labeled with the one or more speaker representations and a speaker location within the acoustic environment.

A method of incorporating environmental acoustic sources into a virtual environment by measuring real environment acoustic sources and locations and incorporating them into a virtual environment with virtual acoustic sources.

An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

An electronic device having circuitry, which is configured to estimate a distraction level of an audio object stream, and to modify the audio object stream based on the estimated distraction level to obtain a modified audio object stream.

An apparatus comprising means including circuitry configured for: applying equalization to a sub-set of a plurality of spatial sound sources, included in a sound scene, to modify the sound scene, wherein spatial sound sources are associated with respective locations in the sound scene, wherein equalization includes frequency-dependent level adaptation, and wherein the sub-set includes multiple spatial sound sources but does not include all of the plurality of spatial sound sources.

An object of the present disclosure is to provide a filter generation device and a filter generation method, capable of generating a filter suitable for out-of-head localization processing. A processing device according to an embodiment includes: a frequency characteristics acquisition unit configured to acquire frequency characteristics based on sound pickup signals; a level calculation unit configured to calculate a reference level in the frequency characteristics; a correction unit configured to correct the frequency characteristics so that the frequency characteristics fall within a predetermined level range including the reference level, and thereby calculate corrected characteristics; and a filter generation unit configured to generate a corrected filter based on the corrected characteristics.