Abstract: A method for producing a multi-channel audio signal from one or more sound object signals is disclosed, where, for each sound object signal a plurality of width signals is produced, the amplitudes of the width signals following a substantially Gaussian distribution. The width signals are processed to produce a plurality of pan signals which are mapped to at least one channel. Each channel in the audio signal is produced by combining the pan signals from each sound object. An apparatus for reproducing such a multi-channel audio signal is also disclosed, in which a plurality of first loudspeakers are provided spaced around a first arc forward of a predetermined listening zone, each of the first loudspeakers facing towards the listening zone and substantially equidistant therefrom. A plurality of second loudspeakers are provided spaced around a second arc behind the listening zone, each of the second loudspeakers facing towards the listening zone.

Abstract: Embodiments relate to obtaining head-related transfer function (HRTF) through performing simulation using images of a user's head. The geometry of the user's head is determined based in part on one or more images of the user's head. The simulation of sound propagation from an audio source to the user's head is performed based on the generated geometry. The geometry may be represented in a three-dimensional meshes or principal component analysis (PCA)-based where the user's head is represented as a combination of representative three-dimensional shapes of test subjects' heads.

Abstract: A system is provided for streaming media content in a vehicle. The system includes a personal media streaming appliance system configured to connect to a media delivery system and receive media content from the media delivery system at least via a cellular network. The personal media streaming appliance system operates to transmit a media signal representative to the received media content to a vehicle media playback system so that the vehicle media playback system operates to play the media content in the vehicle.

Abstract: A method for increasing stability of an inter-channel time difference (ICTD) parameter in parametric audio coding, wherein a multi-channel audio input signal comprising at least two channels is received. The method comprises obtaining an ICTD estimate, ICTDest(m), for an audio frame m and a stability estimate of said ICTD estimate, and determining whether the obtained ICTD estimate, ICTDest(m), is valid. If the ICTDest(m) is not found valid, and a determined sufficient number of valid ICTD estimates have been found in preceding frames, a hang-over time is determined using the stability estimate and a previously obtained valid ICTD parameter, ICTD (m?1), is selected as an output parameter, ICTD (m), during the hang-over time. The output parameter, ICTD (m), is set to zero if valid ICTDest(m) is not found during the hang-over time.

Abstract: Various embodiments relate generally to electrical and electronic hardware, computer software, wired and wireless network communications, and audio and speaker systems. More specifically, disclosed are an apparatus and a method for processing signals for optimizing audio, such as 3D audio, by adjusting the filtering for cross-talk cancellation based on listener position and/or orientation. In one embodiment, an apparatus is configured to include a plurality of transducers, a memory, and a processor configured to execute instructions to determine a physical characteristic of a listener relative to the origination of the multiple channels of audio, to cancel crosstalk in a spatial region coincident with the listener at a first location, to detect a change in the physical characteristic of the listener, and to adjust the cancellation of crosstalk responsive to detecting the change in the physical characteristic to establish another spatial region at a second location.

Abstract: At least one acoustic parameter of an environment is obtained by emitting a sound signal emitted, receiving a sound signal received, estimating a sampling frequency shift and of a temporal shift between the emission and reception devices on the basis of impulse responses obtained on the basis of portions of the sound signal received and of corresponding portions of the sound signal emitted, obtaining a corrected sound signal received by applying to the sound signal received a sampling frequency shift correction and a temporal shift correction on the basis of the estimating, determining the impulse response of the environment on the basis of portions of the corrected sound signal received and of corresponding portions of the sound signal emitted, and obtaining acoustic parameters of the environment on the basis of the impulse response.

Abstract: Headphones provide binaural sound to a location behind a portable electronic device. The sound is convolved or processed to a location that is behind the portable electronic device so that the listener perceives the location of the sound as originating from the location of the portable electronic device.

Abstract: An audio instrument includes a component assembly, a pan-tilt mechanism, and one or more processors. The component assembly includes a parabolic reflector, an audio device, and a sensor unit. The audio device is configured to one or more of receive or transmit sound waves via a transducer element disposed within a communication envelope of the parabolic reflector. The sensor unit includes a distance measurement sensor. The pan-tilt mechanism orients the component assembly at independent angular rotations about two orthogonal axes relative to a support platform. The one or more processors are configured to control the pan-tilt mechanism to orient the component assembly such that a central axis of the parabolic reflector is aimed towards a specific three-dimensional location for targeted audio communication between the audio device and a target at the specific three-dimensional location.

Abstract: An audio system with first and second discrete audio drivers that are in a listening space, wherein each driver is configured to radiate sound into the listening space, and wherein the radiated sound from each driver at a listening location in the listening space has a frequency, phase, and magnitude. The audio system is configured to modify a phase of the sound radiated by at least one of the first or second drivers such that a phase response difference between the first and second drivers at the listening location is closer to zero degrees than it was before the modification.

Abstract: A dry audio signal may be received. An envelope of the dry audio signal may be compared to a threshold of the playback device to determine a dry excess. A filter may be applied to the dry audio signal to generate a wet audio signal. An envelope of the wet audio signal may be compared to the threshold to determine a wet excess. Based on the dry and wet excess, a dry gain may be determined for the dry audio signal and a wet gain may be determined for the wet audio signal. The wet audio signal and the dry audio signals may be mixed in proportion to respective gains. The mixed audio signal may avoid exceeding an operational limit associated with audio playback on the playback device.

Abstract: An audio system captures audio data of test sounds through a microphone of a headset worn by a user. The test sounds are played by an external speaker, and the audio data includes audio data captured for different orientations of the headset with respect to the external speaker. A set of head-related transfer function (HRTFs) is calculated based at least in part on the audio data of the test sounds at the different orientations of the headset. A portion of the set of HRTFs is discarded to create an intermediate set of HRTFs. The discarded portion corresponding to one or more distortion regions that are based in part on wearing the headset. One or more HRTFs are generated that correspond to the discarded portion using at least some of the intermediate set of HRTFs to create an individualized set of HRTFs for the user.

Abstract: A method for securing a computer system comprising a step of: disguising a password entry screen.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: A processor in an audio extraction apparatus performs a preprocessing operation to determine, for a stereo audio source including first channel audio data including an accompaniment sound and a vocal sound for a first channel and second channel audio data including an accompaniment sound and a vocal sound for a second channel, a difference between the first channel audio data and the second channel audio data to generate center cut audio data, and an audio extraction operation to input the first channel audio data, the second channel audio data and the center cut audio data to a trained machine learning model to extract any one of the accompaniment sound and the vocal sound.

Abstract: Methods and apparatus improve a user experience localizing binaural sound to an augmented reality (AR) or virtual reality (VR) image. The sound is convolved or processed to a location that is behind the AR or VR image so that the listener perceives the location of the sound as originating from the AR or VR image.

Abstract: The present disclosure relates to blended analog-to-digital conversion for digital test and measurement devices. A first analog-to-digital converter (ADC) converts an analog signal into a first digital signal a first sampling rate. A digital filtering component generates a filtered digital signal by processing the first digital signal. An analog low pass filter filters the analog signal to generate a filtered analog signal. A second ADC converts the filtered analog signal into a second digital signal. A digital subtractor circuit subtracts the filtered digital signal from the first digital signal or the second digital signal. A digital adder circuit generates a blended digital signal by processing an output of the digital subtractor circuit and one of the first digital signal or the second digital signal.

Abstract: In some examples, crosstalk cancellation for speaker-based spatial rendering may include perceptually smoothing head-related transfer functions (HRTFs) corresponding to ipsilateral and contralateral transfer paths of sound emitted from first and second speakers to corresponding first and second destinations. The crosstalk cancellation may further include inserting an inter-aural time difference in the perceptually smoothed HRTFs corresponding to the contralateral transfer paths. A crosstalk canceller may be generated by inverting the perceptually smoothed HRTFs corresponding to the ipsilateral transfer paths and the perceptually smoothed HRTFs corresponding to the contralateral transfer paths including the inserted inter-aural time difference.

Abstract: A first playback stream presentation intended for reproduction on a first audio reproduction system and transform parameters may be received and decoded. The second playback stream presentation may be intended for reproduction on headphones. The transform parameters may be applied to an intermediate playback stream presentation to obtain the second playback stream presentation. The intermediate playback stream presentation may be the first playback stream presentation, a downmix of the first playback stream presentation, or an upmix of the first playback stream presentation. A cross-talk-cancelled signal may be obtained by processing the second playback stream presentation with a cross-talk cancellation algorithm. The cross-talk-cancelled signal may be processed by a dynamic equalization or gain stage wherein an amount of equalization or gain may be dependent on a level of the first playback stream presentation or the second playback stream presentation.

Abstract: An apparatus for encoding a multi-channel signal having at least three channels includes an iteration processor, a channel encoder and an output interface. The iteration processor is configured to calculate inter-channel correlation values between each pair of the at least three channels, for selecting a pair including a highest value or including a value above a threshold, and for processing the selected pair using a multi-channel processing operation to derive first multi-channel parameters for the selected pair and to derive first processed channels. The iteration processor is configured to perform the calculating, the selecting and the processing using at least one of the processed channels to derive second multi-channel parameters and second processed channels. The channel encoder is configured to encode channels resulting from an iteration processing to obtain encoded channels.

Abstract: Embodiments relate to b-chain processing for a spatially enhanced audio signal. A system includes a b-chain processor. The b-chain processor determines asymmetries between the left speaker and the right speaker in frequency response, time alignment, and signal level for a listening position; and generates a left output channel for the left speaker and a right output channel for the right speaker by: applying an N-band equalization to the spatially enhanced signal to adjust for the asymmetry in the frequency response; applying a delay to the spatially enhanced signal to adjust for the asymmetry in the time alignment; and applying a gain to the spatially enhanced signal to adjust for the asymmetry in the signal level.

Abstract: A speaker system is provided with a docking station, a removable housing mounted to the docking station, and a transducer supported by the removable housing. The speaker system is also provided with a processor supported by the removable housing and programmed to separate an audio signal into channels and provide at least one channel to the transducer based a location of the removable housing relative to the docking station.

Abstract: A method of generating binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database, wherein the multiple audio source signals can be channel-based, object-based, or a mixture of both signals. The method includes grouping the multiple audio source signals according to positions of the audio sources in a hierarchical manner, and parameterizing BRIR to be used for rendering. The method also includes dividing each audio source signal to be rendered into a number of blocks and frames, averaging the parameterized BRIR sequences identified with a hierarchically grouping result, and downmixing the divided audio source signals identified with the hierarchically grouping result.

Abstract: Embodiments herein are primarily described in the context of a system, a method, and a non-transitory computer readable medium for producing a sound with enhanced spatial detectability and reduced crosstalk interference. The audio processing system receives an input audio signal, and performs an audio processing on the input audio signal to generate an output audio signal. In one aspect of the disclosed embodiments, the audio processing system divides the input audio signal into different frequency bands, and enhances a spatial component of the input audio signal with respect to a nonspatial component of the input audio signal for each frequency band.

Abstract: An audio processing apparatus has a setting processor that sets a size of a virtual sound source; and a signal processor that generates an audio signal by imparting to an audio signal a plurality of head-related transfer characteristics. The plurality of head-related transfer characteristics corresponds to respective points within a range that accords with the size set by the setting processor from among a plurality of points, with each point having a different position relative to a listening point.

Abstract: According to an example aspect of the present invention, there is provided a method for forming a binaural filter for a stereo headphone in order to preserve the sound quality of the headphone, whereby the sum of the direct and crosstalk paths from loudspeakers to each ear have flat magnitude responses.

Abstract: Embodiments are described for a soundfield system that receives a transmitting soundfield, wherein the transmitting soundfield includes a sound source at a location in the transmitting soundfield. The system determines a rotation angle for rotating the transmitting soundfield based on a desired location for the sound source. The transmitting soundfield is rotated by the determined angle and the system obtains a listener's soundfield based on the rotated transmitting soundfield. The listener's soundfield is transmitted for rendering to a listener.

Abstract: There is provided a local sound field forming apparatus including a local sound field forming filter coefficient recording unit that records an audio filter coefficient for forming a sound field by an evanescent wave, a filter unit that convolves the audio filter coefficient and a sound source signal to generate a loudspeaker drive signal, and a loudspeaker array that includes a plurality of loudspeakers including a directional loudspeaker, and that reproduces a sound on the basis of the loudspeaker drive signal. The present technology can be applied to a local sound field forming apparatus.

Abstract: A system for producing an encoded digital audio recording has an audio encoder that encodes a digital audio recording having a number of audio channels or audio objects. An equalization (EQ) value generator produces a sequence of EQ values which define EQ filtering that is to be applied when decoding the encoded digital audio recording, wherein the EQ filtering is to be applied to a group of one or more of the audio channels or audio objects of the recording independent of any downmix. A bitstream multiplexer combines the encoded digital audio recording with the sequence of EQ values, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording.

Abstract: The present technology relates to an acoustic signal processing apparatus, an acoustic signal processing method and a program which can widen the variations of the configuration of a virtual surround system that stabilizes the localization sensation of a virtual speaker. Crosstalk correction processing is performed on a first binaural signal based on a sound source opposite side HRTF and a second binaural signal based on a sound source side HRTF. A first acoustic signal and a second acoustic signal are generated. A component of a first frequency band, in which a first notch of the sound source opposite side HRTF appears, and a component of a second frequency band, in which a second notch appears, are attenuated in an input signal or the second binaural signal, thereby attenuating the component of the first frequency band and the component of the second frequency band of the first acoustic signal and the second acoustic signal.

Abstract: A control unit for an audio system including a subwoofer and at least one further loudspeaker. Type information is recorded via the at least one connected loudspeaker, and a configurable filter for the at least one connected loudspeaker is configured by means of appurtenant configuration information from a memory. The configurable filter is configured so that a base phase management is provided for the transition frequency range between the subwoofer and the at least one connected loudspeaker so that the phase of the sound signals of the subwoofer is matched to the phase of the sound signals of the at least one connected loudspeaker.

Abstract: A method of reproducing a multi-channel audio signal including an elevation sound signal in a horizontal layout environment is provided, thereby obtaining a rendering parameter according to a rendering type and configuring a down-mix matrix, and thus effective rendering performance may be obtained with respect to an audio signal that is not suitable for applying virtual rendering. A method of rendering an audio signal includes receiving a multi-channel signal includes a plurality of input channels to be converted into a plurality of output channels; determining a rendering type for elevation rendering based on a parameter determined from a characteristic of the multi-channel signal; and rendering at least one height input channel according to the determined rendering type, wherein the parameter is included in a bitstream of the multi-channel signal.

Abstract: Embodiments relate to providing a conference for client devices with spatialized audio. Input audio streams are received from the client devices. For each client device, placement data defining spatial locations of other client devices within a sound field is determined. A mixed stream including a left mixed channel and a right mixed channel for the client device is generated by mixing and panning input audio streams of the other client devices according to the placement data. A spatially enhanced stream including a left enhanced channel for a left speaker and a right enhanced channel for a right speaker is generated by applying subband spatial processing and crosstalk processing on the left mixed channel and the right mixed channel of the mixed stream.

Abstract: A method is provided for directing a user to a point of interest using sounds played on a wearable audio output device. A user's direction of gaze is determined using data from at least one sensor in the audio output device. First and second locations are determined of the user and the point of interest respectively, and a distance is determined between the first and second locations. A relative angle between the direction of gaze and the direction of the point of interest is determined based on the determined first and second locations. A sample sound is altered using at least one sonification technique to convey information relating to the determined distance and the relative angle. The altered sample sound is rendered and output for playing by the audio output device.

Abstract: A computer-implemented method for reducing undesired crosstalk signals on an inactive channel of a device comprising the steps of: (i) determining system volume levels required to achieve a range of desired audio output attenuation levels on an active channel of the device; (ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude and associated phase shift required to reduce the undesired crosstalk on the ina

Abstract: A signal generator has a filter bank that provides weighted versions of audio signals to speakers. The weights were derived by identifying a first constraint that limits a weight that can be applied to an audio signal to be provided to a first speaker. A characteristic of a second speaker that affects how a user will perceive audio signals output by that speaker relative to audio signals output by the first speaker was also determined. A second constraint was determined based on the determined characteristic and the first constraint. The weights were then determined so as to minimize a difference between an actual balance of each signal that is expected to be heard by a user and a target balance. The signal generator can achieve sweet spot correction and sound stage widening simultaneously. It also achieves a balanced sound stage, particularly when the speakers are asymmetric.

Abstract: A method of processing audio data for replay on a mobile device with a first speaker and a second speaker, wherein the audio data comprises a respective audio signal for each of the first and second speakers, includes: determining a device orientation of the mobile device; if the determined device orientation is vertical orientation, applying a first processing mode to the audio signals for the first and second speakers; and if the determined device orientation is horizontal orientation, applying a second processing mode to the audio signals for the first and second speakers.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: A system is described herein that comprises a collar device including a sound masking component, the collar device including one or more collar device sensors for detecting physiological data of an animal. The system includes one or more environmental sensors configured to detect environmental data of the animal's environment and to transmit the environmental data to the collar device. The collar device comprises one or more applications running on at least one processor for detecting an occurrence of one or more events using at least one of the physiological data, the environmental data, and outcome data. The one or more applications are configured to use information of the one or more events to select a sound masking signal for delivery after the occurrence of the one or more events The system includes a sound masking component for delivering the sound masking signal.

Abstract: A three-dimensional sound reproducing method includes: acquiring a multichannel audio signal; rendering signals to a channel to be reproduced according to channel information and a frequency of the multichannel audio signal; and mixing the rendered signals.

Abstract: A sound control method includes altering at least one characteristic of power delivered within an electrified vehicle powertrain to cause at least one component of the electrified vehicle powertrain to emit different acoustic tones that follow a predetermined sequence. A sound control assembly includes a power characteristic control system that alters at least one characteristic of power delivered within an electrified vehicle powertrain to cause at least one component of the electrified vehicle powertrain to emit different acoustic tones that follow a predetermined sequence.

Abstract: In some examples, matrix decomposition of audio signal processing filters for spatial rendering may include determining first and second spatial synthesis filters respectively as a sum and a difference of ipsilateral and contralateral spatial synthesis filters, and determining first and second crosstalk cancellation filters respectively as a sum and a difference of ipsilateral and contralateral crosstalk cancellation filters. A combined spatial synthesizer and crosstalk canceller that includes a first combined filter and a second combined filter may be determined based on application of matrix decomposition to the first and second spatial synthesis filters and the first and second crosstalk cancellation filters. Further, spatial synthesis and crosstalk cancellation on first and second input audio signals may be performed based on application of the combined spatial synthesizer and crosstalk canceller.

Abstract: An audio processing method includes: providing an input audio signal; performing a transformation process to transform the input audio signal from a time domain to a frequency domain; performing a panning process on an amplitude spectrum corresponding to the input audio signal to obtain a panning amplitude signal; performing a first broader process and a second broader process on a phase spectrum corresponding to the input audio signal to obtain a left channel separation phase signal and a right channel separation phase signal; performing a first inverse transformation process on the panning amplitude signal and the left channel separation phase signal and performing a second inverse transformation process on the panning amplitude signal and the right channel separation phase signal to obtain an optimized left channel output audio signal and an optimized right channel output audio signal corresponding to the time domain.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by rendering an audio sound field representation for arbitrary spatial loudspeaker setups and/or by a decoder that decodes based on a decode matrix (D). The decode matrix (D) is based on smoothing and scaling of a first decode matrix {circumflex over (D)} with smoothing coefficients. The first decode matrix {circumflex over (D)} is based on a mix matrix G and a mode matrix {tilde over (?)}, where the mix matrix G was determined based on L speakers and positions of a spherical modelling grid related to a HOA order N, and the mode matrix {tilde over (?)} was determined based on the spherical modelling grid and the HOA order N.

Abstract: A device includes a processor that is configured to determine an inter-channel mismatch value indicative of a temporal misalignment between a frequency-domain reference channel and a frequency-domain target channel. The processor is also configured to adjust the frequency-domain target channel based on the inter-channel mismatch value to generate an adjusted frequency-domain target channel. The processor is further configured to perform a down-mix operation, based on the frequency-domain reference channel and the adjusted frequency-domain target channel, to generate a mid channel and a side channel. The processor is also configured to generate a predicted side channel based on the mid channel. The processor is further configured to generate a residual channel based on the side channel and the predicted side channel. The processor is also configured to encode the residual channel as part of a bitstream.

Abstract: Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer ?2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer ?3.

Abstract: An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by an author of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are each set for corresponding playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data in the second format.

Abstract: An audio processing apparatus includes a receiver configured to receive audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers. A renderer is configured to generate audio transducer signals for the set of audio transducers from the audio data, and to render audio components in accordance with rendering modes. A render controller is configured to select a rendering mode for the renderer based on the audio transducer position data. The renderer is configured to employ different rendering modes for different subsets of the set of audio transducers and the render controller is configured to independently select rendering modes for each of the different subsets of the set of audio transducer including selecting the rendering mode for a first audio transducer in response to a position of the first audio transducer relative to a predetermined position for the first audio transducer.

Abstract: Circuitry can separate a multi-channel input signal into a center signal and a residual signal, apply time-varying center and residual gains to the center and residual signals, and combine the gain-adjusted center and residual signals to form a multi-channel audio signal. The center and residual gains are automatically determined in response to the center and residual signals so as to prevent each channel of the multi-channel audio signal from exceeding a target volume. During first times, the center and residual gains can vary synchronously so as to ensure that amplitude and phase relationships among channels in the multi-channel input signal are retained into the multi-channel audio signal. During second times, the center and residual gains can vary independently so as to reduce the energy of the residual signal compared to the center signal in the multi-channel audio signal.

Abstract: A signal processor that performs crosstalk cancellation on an audio signal that is input, in a distorted acoustic space where two speakers are placed, the two speakers including an X-side speaker and a Y-side speaker, where X denotes one of left and right, and Y denotes the other of left and right. The signal processor causes the Y-side speaker to output a sound of the audio signal and causes the X-side speaker to output a sound of a signal obtained by processing the audio signal using transfer function GCY, where a transfer function between the Y-side speaker and the Y-side ear is defined as GYY, a transfer function between the X-side speaker and the Y-side ear is defined as GXY, and a transfer function obtained by dividing the transfer function GYY by the transfer function GXY is defined as GCY.

Abstract: The invention relates to a method for processing customized data representative of the directivity of an customized audio system, the method comprising the following steps: —obtaining (101), for each individual of an initial set of individuals, at least one customized suite of filters; —determining (103) N independent components common to the suites of filters obtained; —decomposing (104) each of the suites obtained into a first base constructed from the N independent components with a view to obtaining, for each suite of filters, a first suite of weighting coefficients; —decomposing (105) each first suite of weighting coefficients into a second base of P independent components, so as to obtain a second suite of weighting coefficients; —storing (106) each second suite of weighting coefficients obtained in association with an identifier of an individual from among the initial set of individuals.