An audio signal processing method and apparatus for adaptively adjusting a decorrelator. The method comprises obtaining a control parameter and calculating mean and variation of the control parameter. Ratio of the variation and mean of the control parameter is calculated, and a decorrelation parameter is calculated based on the said ratio. The decorrelation parameter is then provided to a decorrelator.

The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

A filter determination system, a filter determination device, a filter determination method and a program with which a filter can be appropriately determined are provided. A processing device according to this embodiment includes a frequency response acquisition unit that acquires a frequency response of a filter for an audio signal, a smoothing unit that smoothes the frequency response and obtains a smoothed response, a candidate point determination unit that determines candidate split points based on a bottom position of the smoothed response, and a split point determination unit that determines one or more band split points from the candidate split points.

An audio processing method includes: 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.

An acoustic signal processing device 1 includes: a focal point position determination unit 12 that obtains a plurality of sets of initial focal point coordinates, coordinates of the virtual sound source, and a direction of directivity thereof, and for a pair of sets of initial focal point coordinates with different polarities among the plurality of sets of initial focal point coordinates, multiplies the sets of initial focal point coordinates by a rotation matrix based on the coordinates of the virtual sound source to thereby determine sets of focal point coordinates, the rotation matrix being specified from the direction of the directivity; a circular harmonic coefficient conversion unit 13 that calculates weights to be applied to multipoles including the sets of focal point coordinates from a circular harmonic coefficient; a filter coefficient computation unit 14 that, for each of the speakers in the speaker array, computes a weighted driving function to be applied to the speaker from the sets of focal point coordinates, polarities of the sets of focal point coordinates, and the weights to be applied to the multipoles; and a convolutional operation unit 15 that, for each of the speakers in the speaker array, convolves the weighted driving function for the speaker into the input acoustic signal to output the output acoustic signal for the speaker.

In device having at least one microphone and one or more speakers, environmental sound may be recorded using the microphone, classified and mixed with source media sound to produce a mixed sound depending on the classification. The mixed sound may then be played over the one or more speakers.

In some examples, an audio control system can include a first set of resources, a second set of resources and a controller. The first set of resources can generate a frequency energy band representation of a multi-channel source audio input. Additionally, the second set of resources can determine at least a value representing a strength of correlation between multiple channels of the multi-channel source audio input. Moreover, the audio output controller can determine a set of control parameters for tuning sound creation from an audio signal generator to reflect a set of spatial characteristics of the source audio input, based on the frequency energy band representation and the first value.

An audio system provides for spatial enhancement, crosstalk processing, and crosstalk compensation of an input audio signal. The crosstalk compensation compensates for spectral defects caused by the application of the crosstalk processing to a spatially enhanced signal. The crosstalk compensation may be performed prior to the crosstalk processing, after the crosstalk processing, or in parallel with the crosstalk processing. The crosstalk compensation includes applying filters to the mid and side components of the left and right input channels to compensate for spectral defects from crosstalk processing of the audio signal. The crosstalk processing may include crosstalk simulation or crosstalk cancellation. In some embodiments, the crosstalk compensation may be integrated with a subband spatial processing that spatially enhances the audio signal.

An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor $g=\frac{1}{\sqrt{L}}.$
The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.