An acoustic system can include an acoustic processor that is configured to analyze and output acoustic signals to each of first and second loudspeakers corresponding to each of first and second acoustic zones of a shared acoustic space. The system can determine a first measure of loudness associated with a first acoustic signal. The system can be configured to output the first acoustic signal as sound on the first loudspeaker in the first acoustic zone. The system can determine a second measure of loudness associated with a second acoustic signal. The system can be configured to output the second acoustic signal as sound on the second loudspeaker in the second acoustic zone. The system can also modify the second acoustic signal based on the first acoustic signal.

An acoustic system can include an acoustic processor that is configured to analyze and output acoustic signals to each of first and second loudspeakers corresponding to each of first and second acoustic zones of a shared acoustic space. The system can determine a first measure of loudness associated with a first acoustic signal. The system can be configured to output the first acoustic signal as sound on the first loudspeaker in the first acoustic zone. The system can determine a second measure of loudness associated with a second acoustic signal. The system can be configured to output the second acoustic signal as sound on the second loudspeaker in the second acoustic zone. The system can also modify the second acoustic signal based on the first acoustic signal.

A sound processing system includes a sound input device for providing a sound input, a sound output device for providing a sound output, and processing electronics including a processor and a memory, wherein the processing electronics is configured to receive a target sound input identifying a target sound, receive a rule input establishing a sound processing rule that references the target sound, receive a sound input from the sound input device, analyze the sound input for the target sound, process the sound input according to the sound processing rule in view of the analysis of the sound input, and provide a processed sound output to the sound output device.

There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M≦N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

A method for calibrating a distributed audio reproduction system, including a set of N heterogeneous loudspeakers controlled by a server. This method includes the following steps: a) placing a microphone in front of a first loudspeaker of the set; b) capturing), by the microphone, a calibration signal sent to the loudspeaker at a first time and reproduced by same; c) capturing, by the microphone, the calibration signal sent with a known time delay to the N−1 other loudspeakers of the set and reproduced by these N−1 loudspeakers; d) capturing, by the microphone, the calibration signal sent to the first loudspeaker at a second time and reproduced again by same; e) repeating steps a) to d) for the N loudspeakers of the set; f) determining a plurality of heterogeneity factors to be corrected for the set of N loudspeakers by analysing the data thus captured; g) correcting the determined heterogeneity factors.

A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.

One embodiment provides a method of automatic spatial calibration. The method comprises estimating one or more distances from one or more loudspeakers to a listening area based on a machine learning model and one or more propagation delays from the one or more loudspeakers to the listening area. The method further comprises estimating one or more incidence angles of the one or more loudspeakers relative to the listening area based on the one or more propagation delays. The method further comprises applying spatial perception correction to audio reproduced by the one or more loudspeakers based on the one or more distances and the one or more incidence angles. The spatial perception correction comprises delay and gain compensation that corrects misplacement of any of the one or more loudspeakers relative to the listening area.

In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

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.