Abstract: The present document relates to methods and apparatus for rendering input audio for playback in a playback environment. The input audio includes at least one audio object and associated metadata, and the associated metadata indicates at least a location of the audio object. A method for rendering input audio including divergence metadata for playback in a playback environment comprises creating two additional audio objects associated with the audio object such that respective locations of the two additional audio objects are evenly spaced from the location of the audio object, on opposite sides of the location of the audio object when seen from an intended listener's position in the playback environment, determining respective weight factors for application to the audio object and the two additional audio objects, and rendering the audio object and the two additional audio objects to one or more speaker feeds in accordance with the determined weight factors.

Abstract: A system for rotating sound or selective listening to sound provides for the ability of the apparent direction of sound sources in a listening environment to remain in consistent orientations in space despite rotations of the microphones used to capture the sound and despite rotations of the head of the listener, even when wearing headphones. Modules are provided in the system to distinguish the sound sources and their apparent directions, as well as to optionally rotate the sound sources in response to detected rotations of the listener's head and/or detected rotations of the microphones.

Abstract: The disclosure relates to an audio processing apparatus, comprising: a plurality of audio sensors, each audio sensor configured to receive a respective plurality of audio frames of an audio signal from an audio source, wherein the respective plurality of audio frames defines an audio channel of the audio signal; and a processing circuitry configured to: determine a respective feature set having at least one feature for each audio frame of each of the plurality of audio frames, wherein the plurality of features define a three-dimensional feature array; process the three-dimensional feature array using a neural network, wherein the neural network comprises a self-attention layer configured to process a plurality of two-dimensional sub-arrays of the three-dimensional feature array; and generate an output signal on the basis of the plurality of processed two-dimensional sub-arrays. Moreover, the disclosure relates to a corresponding audio processing method.

Abstract: Methods of operating a system of nodes are provided. A method of operating first and second nodes that are in a system of nodes includes processing first data from a first plurality of sensors of the first node. The method includes controlling a first digital display of the first node in response to the processed first data. The method includes uploading the processed first data from the first node via the Internet. The method includes processing second data from a second plurality of sensors of the second node. The method includes controlling a second digital display of the second node in response to the processed second data. The method includes uploading the processed second data from the second node via the Internet. Each of the first and second nodes is attached to a utility pole, a light pole, a kiosk, or a motor vehicle. Related systems are also provided.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Embodiments are directed to a speaker system that contains an array of multiple dispersion drivers that creates an expansive acoustic pattern to playback multi-channel audio content through a standalone speaker. The speaker system comprises an interface receiving stereo audio; an upmixer generating surround sound formatted audio from the stereo audio including one or more height channels; a virtualizer/downmixer component coupled to the upmixer and generating speaker feeds for two or more loudspeaker output sections, configured to play back the stereo audio, wherein each output section is further configured to play its own dedicated stereo audio signals; and a set of drivers each coupled to a respective output section and configured to project sound in at least two different dispersion patterns.

Abstract: Disclosed herein are systems and methods for presenting audio content in mixed reality environments. A method may include receiving a first input from an application program; in response to receiving the first input, receiving, via a first service, an encoded audio stream; generating, via the first service, a decoded audio stream based on the encoded audio stream; receiving, via a second service, the decoded audio stream; receiving a second input from one or more sensors of a wearable head device; receiving, via the second service, a third input from the application program, wherein the third input corresponds to a position of one or more virtual speakers; generating, via the second service, a spatialized audio stream based on the decoded audio stream, the second input, and the third input; presenting, via one or more speakers of the wearable head device, the spatialized audio stream.

Abstract: There is disclosed inter alia an apparatus for spatial audio signal encoding comprising means for receiving for each time frequency block of a sub band of an audio frame a spatial audio parameter comprising an azimuth and an elevation; determining a first distortion measure for the audio frame by determining a first distance measure for each time frequency block and summing the first distance measure for each time frequency block; determining a second distortion measure for the audio frame by determining a second distance measure for each time frequency block and summing the second distance measure for each time frequency block, and selecting either the first quantization scheme or the second quantization scheme for quantising the elevation and the azimuth for all time frequency blocks of the sub band of the audio frame, wherein the selecting is dependent on the first and second distortion measures.

Abstract: A loudspeaker system can include a first loudspeaker driver provided in a substantially fixed spatial relationship relative to a microphone. The loudspeaker driver can be tuned, for example automatically and without user input. In an example, the tuning can include receiving transfer function reference information about the first loudspeaker driver and the microphone, and receiving information about a desired acoustic response for the loudspeaker system. The tuning can include determining a simulated response for the loudspeaker system using a first input signal and the transfer function reference information, and can include providing the first input signal to the first loudspeaker driver. In response to the first input signal, an actual response for the loudspeaker driver can be received using the microphone. A compensation filter can be determined for the loudspeaker system based on the determined simulated response and the received actual response for the loudspeaker system.

Abstract: An apparatus is provided that includes at least one processor and at least one memory including computer program code with the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to analyze a level difference between a left channel and a right channel of a stereo audio signal and to determine if the level difference between the left channel and the right channel is above a threshold. The apparatus is also caused to conditionally, if the determined level difference is above the threshold, move signal energy from a louder one of the left channel and the right channel to the other of the left channel and the right channel to create a processed left channel and a processed right channel of a processed stereo audio signal. A corresponding method is also provided.

Abstract: Audio data in a first format may be processed to produce audio data in a second format, which may be a reduced or simplified version of the first format. A loudness correction process may produce loudness-corrected audio data in the second format. A first power of the audio data in the second format and a second power of the loudness-corrected audio data in the second format may be determined. A second-format loudness correction factor for the audio data in the second format may be based, at least in part, on a power ratio between the first power and the second power. A first-format loudness correction factor for the audio data in the first format may be based, at least in part, on the power ratio and a power relationship between the audio data in the first format and the audio data in the second format.

Abstract: A signal processing device includes one or more filter processors that perform sound localization processing of an input acoustic signal and generate output signals; a coefficient setter that sets, for the filter processors, a plurality of filter coefficients for use in the filter processors; an output data selector that selects, out of the output signals subjected to the sound localization processing by the filter processors, output signals to be output to a left speaker and a right speaker; a time measurer that monitors a time for switching the output signals; and a controller that causes the output data selector to select the output signals in accordance with the time for switching the output signals. The plurality of filter coefficients include filter coefficients generated under a plurality of hearing conditions.

Abstract: The present technology relates to a signal processing device and method, and a program for improving reproducibility of a sound image with a small amount of calculation. A signal processing device includes a rendering method selection unit configured to select one or more methods of rendering processing of localizing a sound image of an audio signal in a listening space from among a plurality of methods, and a rendering processing unit configured to perform the rendering processing for the audio signal by the method selected by the rendering method selection unit. The present technology can be applied to a signal processing device.

Abstract: An encoding/decoding apparatus and method for controlling a channel signal is disclosed, wherein the encoding apparatus may include an encoder to encode an object signal, a channel signal, and rendering information for the channel signal, and a bit stream generator to generate, as a bit stream, the encoded object signal, the encoded channel signal, and the encoded rendering information for the channel signal.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: Embodiments are described for rendering spatial audio content through a system that is configured to reflect audio off of one or more surfaces of a listening environment. The system includes an array of audio drivers distributed around a room, wherein at least one driver of the array of drivers is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment and a renderer configured to receive and process audio streams and one or more metadata sets that are associated with each of the audio streams and that specify a playback location in the listening environment.

Abstract: A method and apparatus for decompressing a Higher Order Ambisonics (HOA) signal representation is disclosed. The apparatus includes an input interface that receives an encoded directional signal and an encoded ambient signal and an audio decoder that perceptually decodes the encoded directional signal and encoded ambient signal to produce a decoded directional signal and a decoded ambient signal, respectively. The apparatus further includes an extractor for obtaining side information related to the directional signal and an inverse transformer for converting the decoded ambient signal from a spatial domain to an HOA domain representation of the ambient signal. The apparatus also includes a synthesizer for recomposing a Higher Order Ambisonics (HOA) signal from the HOA domain representation of the ambient signal and the decoded directional signal. The side information includes a direction of the directional signal selected from a set of uniformly spaced directions.

Abstract: 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.

Abstract: Example techniques involve outputting multiple audio channels using a multiple driver playback device. An example playback device receives a first and second channel of audio content. The playback device plays back play back the first channel via a first group of audio transducers such that the first group of audio transducers form, via superposition, a first response lobe having a maximum in a first direction. Further, the playback device plays back the second channel via a second group of audio transducers such that the second group of audio transducers form, via superposition, a second response lobe having a maximum in a second direction that is separated by an angle of at least 45° from the first direction.

Abstract: The invention relates to a method for reproducing audio in a multi-channel sound system including two input signals (L and R), wherein output signals are generated for different sound perception levels. In order to develop said method in such a way that audio can be reproduced within a larger range of applications in a multi-channel sound system, according to the invention, only a lower sound perception level (7) and a higher sound perception level (6) are generated, and a maximum of six output signals are generated, a maximum of two output signals being allocated to the lower sound perception level (7) and a maximum of four output signals being allocated to the higher sound perception level (6).

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: An audio processing apparatus includes a sensor configured to output a detection signal in accordance with an orientation of the sensor; a memory storing instructions; and at least one processor that implements the instructions to: sequentially generate, based on the detection signal, orientation information pieces each indicative of the orientation of the sensor; correct a current orientation information piece based on an average of a first plurality of orientation information pieces, among the sequentially generated orientation information pieces, and generate a corrected current orientation information piece; determine a head-related-transfer function in accordance with the corrected current orientation information piece; and apply a sound-image-localization processing to an audio signal based on the determined head-related-transfer function.

Abstract: A system for rotating sound provides for the ability of the apparent direction of sound sources in a listening environment to remain in consistent orientations in space despite rotations of the microphones used to capture the sound and despite rotations of the head of the listener, even when wearing headphones. Modules are provided in the system to distinguish the sound sources and their apparent directions, as well as to rotate the sound sources in response to detected rotations of the listener's head and/or detected rotations of the microphones.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: An audio signal processing system can be configured to provide virtualized audio information in a three-dimensional soundfield using at least a pair of loudspeakers or headphones. The system can include an audio input configured to receive audio program information that includes at least N discrete audio signals, a first virtualization processor circuit configured to generate intermediate virtualized audio information by filtering M of the N audio signals, and a second virtualization processor circuit configured to generate further virtualized audio information by differently filtering K of the N audio signals, wherein K, M, and N are integers. The system can include an audio signal combination circuit to combine the intermediate virtualized audio information with at least one of the N audio signals, other than the M audio signals, to render fewer than N audio signals for transmission to a second virtualization processor circuit.

Abstract: In a particular aspect, an audio processing device includes a position predictor configured to determine predicted position data based on position data. The audio processing device further includes a processor configured to generate an output spatialized audio signal based on the predicted position data.

Abstract: The present technology relates to a signal processing apparatus and method capable of reducing calculation loads, as well as a program. A signal processing apparatus includes an ambisonic gain calculation unit configured to find, on the basis of spread information of an object, an ambisonic gain while the object is present at a predetermined position. The present technology is applicable to an encoder and a decoder.

Abstract: The present disclosure discloses a scene audio effect controlling method and apparatus. The method includes: setting an intersection area between a first scene and a second scene, where the intersection area includes a first parameter control line for performing audio effect control on the first scene and a second parameter control line for performing audio effect control on the second scene; determining a position parameter of a game role in the intersection area; determining, respectively on the first parameter control line and the second parameter control line, a first scene audio effect control parameter and a second scene audio effect control parameter that correspond to the position parameter; and setting a first scene audio effect according to the first scene audio effect control parameter and setting a second scene audio effect according to the second scene audio effect control parameter.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: A method of estimating an individualized head-related transfer function and an individualized interaural time difference function of a particular person, comprises the steps of: a) obtaining a plurality of data sets comprising a left and a right audio sample from in-ear microphones, and orientation information from an orientation unit, measured in a test-arrangement where an acoustic test signal is rendered via a loudspeaker and the person is moving the head; b) extracting interaural time difference values and/or spectral values, and corresponding orientation values; c) estimating a direction of the loudspeaker relative to the head using a predefined quality criterion; d) estimating an orientation of the orientation unit relative to the head; e) estimating the individualized ITDF and the individualized HRTF. A computer program product may be provided for performing the method, and a data carrier may contain the computer program.

Abstract: A method according to which a group of information sources referenced in a database of containers and information sources is connected. Information containers are displayed on display means and controlled via interface means, implementing a device which allows the operators to perform operations on the information containers. The operations on an information container are recorded by the device in an operations log of a digital processing system, and a processing of the operations log determines relationship links among the information sources. The data are manipulated by the operators intuitively and naturally in the form of containers, and objective relationship links among the information sources are established without requiring subjective interpretation by the operators.

Abstract: A method for acoustic scene playback is described, which comprises: providing recording data comprising microphone signals of microphone setups positioned within an acoustic scene and microphone metadata of the microphone setups, each of the microphone setups has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position within the acoustic scene; assigning each microphone setup Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position.

Abstract: 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.

Abstract: A signal processing method includes: multiplying at least one of M signals output from M microphones by a gain so as to equalize sound pressure levels of the M signals, the M signals representing sounds that arrive at the M microphones from a sound source located within a predetermined distance from the M microphones, M being an integer equal to or greater than two; delaying at least one of the M signals so as to resolve time discrepancies between the M signals, the time discrepancies being caused by differences in arrival time between sounds that arrive at the M microphones from the sound source; and applying a filter so as to suppress a signal which is included in the M signals obtained through the multiplying and the delaying and which represents a sound output from the sound source located within the predetermined distance.

Abstract: A system and method of processing a sound signal. The method of processing a sound signal that includes the following steps: Synchronous reception of an input sound signal (Sinput) of N microphones, N being a natural number greater than or equal to three; Encoding of the input sound signal (Sinput) in a data format (D) of sound, the encoding including a sub-step of transforming the input signal into an ambisonic format of order R, R being a natural number greater than or equal to one, the sub-step of transformation into an ambisonic format is carried out by a Fast Fourier Transform, a matrix multiplication, an Inverse Fast Fourier Transform and by a band pass filter; and Return of an output sound signal (Soutput) by digital processing of the sound data (D).

Abstract: An audio system on a headset presents, to a user, audio content simulating a target artificial reality environment. The system receives audio content from an environment and analyzes the audio content to determine a set of acoustic properties associated with the environment. The audio content may be user generated or ambient sound. After receiving a set of target acoustic properties for a target environment, the system determines a transfer function by comparing the set of acoustic properties and the target environment's acoustic properties. The system adjusts the audio content based on the transfer function and presents the adjusted audio content to the user. The presented adjusted audio content includes one or more of the target acoustic properties for the target environment.

Abstract: The present disclosure relates to a method and audio system for customizing a set of head-related transfer functions (HRTFs) for a user of the audio system to account for the user's bias in hearing. The audio system first presents, via one or more speakers on a headset, audio content to the user wearing the headset, the audio content generated using a set of HRTFs. The audio system monitors responses of the user to the audio content. The audio system customizes the set of HRTFs for the user based on at least one of the monitored responses. The audio system updates audio content using the customized set of HRTFs. The audio system presents the updated audio content to the user with the speakers on the headset.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: A method and system for communicating audio, video, and/or control signals within a home entertainment system. A plurality of audio channels is communicated between a wireless transmitter and a wireless receiver. The wireless transmitter is located proximate to a speaker housing. In some embodiments the speaker housing also encloses a center channel loudspeaker. The center channel loudspeaker transmits an audio signal to a remote loudspeaker. An exemplary remote loudspeaker is a subwoofer loudspeaker. The subwoofer loudspeaker provides one or more received audio channels to one or more surround loudspeakers.

Abstract: The invention relates to a sound wall comprising in total M elements comprising a front face (F), where the sound wall (1) has a lateral extension along an x-axis and a vertical extension along a y-axis, where the sound wall (1) comprises N elements (3), where N?M, which N elements in the front face (F) of each respective of these N elements are provided with a sound radiating unit, such as one or more loudspeaker units, configured to radiate sound energy from the respective front face and into the surroundings, such that a sound field can be created in front of the sound wall; where the sound wall is provided with signal providing means configured to provide the sound radiating units with a signal that is a combination of input signals, such as the left and right channel signals S1(t), S2(t) of a stereophonic signal.

Abstract: 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.

Abstract: Provided are a three-dimensional (3D) sound reproducing method and apparatus. The method includes transmitting sound signals through a head related transfer function (HRTF) corresponding to a first elevation, generating a plurality of sound signals by replicating the filtered sound signals, amplifying or attenuating each of the replicated sound signals based on a gain value corresponding to each of speakers, through which the replicated sound signals will be output, and outputting the amplified or attenuated sound signals through the corresponding speakers.

Abstract: A console for supplying medical-technical devices in a treatment room has a support frame and several connectors or sockets for medical-technical supply or for withdrawal of gases and/or electrical supply currents, as well as for providing electronic communication paths for medical-technical end devices. The console can incorporate an audio playback system integrated in the support frame for filling the treatment room with predetermined soothing or therapeutic sounds. The support system may be adapted for being mounted to or suspended from a ceiling mount in a treatment room, and can have a support frame having an integrated and medically approved audio playback system.

Abstract: A method, apparatus, and computer program product are therefore provided for providing a navigation user interface. The apparatus may be caused to: receive an indication of location based information for a user; provide for generation of a first auditory cue, where generation of the first auditory cue may include generating an audio cue having a virtual source location using three-dimensional audio effects. Providing for generation of the first auditory cue may include causing the apparatus to: provide for generation of a beginning of the first auditory cue at a first virtual source location; and provide for generation of a transition phase of the first auditory cue moving the virtual source location from the first virtual source location along a trajectory and ending at a second virtual source location, where the second virtual source location is a location positioned between the user and a location identified in the location based information.

Abstract: An input audio signal is all-pass filtered and scaled, before being combined with an original version of the input audio signal, to produce an output signal. Envelope of the input signal is detected and the all-pass filter is altered dynamically or in real time, as a function of the detected envelope. Other embodiments are also described and claimed.

Abstract: Example techniques involve balancing bass response of a multiple driver playback device. An example playback device receives left and right channels of audio content and generates a center channel of the audio content by combining at least a portion of the left right channels. The playback device generates first and second side channels of the audio content by combining the center channel and a difference of the left channel and the right channel and combining the center channel and an inverse of the difference of the left channel and the right channel, respectively. The first and second side channels are attenuated by a filter with a given cutoff frequency. The center channel is amplified in proportion to the attenuation of the first side channel. The playback device plays back the first side channel, second side channel, and center channel according to respective radiation patterns having maximums aligned in respective directions.

Abstract: Techniques for using acoustic notifications output via a multi-channel speaker configuration for pedestrian notification are described. Computing device(s) can determine a plurality of signals for dynamically emitting a plurality of sounds via a plurality of speakers associated with an acoustic array disposed on a vehicle. Each signal of the plurality of signals can be emitted from speaker(s) of the plurality of speakers that is/are associated with a particular channel of a plurality of channels. Furthermore, the computing device(s) can cause a portion of the plurality of speakers to dynamically emit the plurality of sounds so that the plurality of sounds are spatialized across the at least one surface of the vehicle. As a result, a pedestrian proximate the vehicle can localize the vehicle as a source of the acoustic notification and perceive a size, or other geometric characteristic, of the vehicle.

Abstract: An audio system includes a first audio apparatus that receives a first audio signal with a first number of channels and a second audio apparatus connected to the first audio apparatus via a network. The first audio apparatus includes a signal processing unit configured to generate, on the basis of the first audio signal received from the outside of the first audio apparatus, a second audio signal with a second number of channels larger than the first number of channels, and a communication unit configured to transmit a third audio signal corresponding to a part of the channels in the second audio signal generated by the signal processing unit to the second audio apparatus.

Abstract: A sound system for a vehicle is provided, which includes as sound sources disposed at given positions of the vehicle, an engine and an audio device for an internal space of a cabin. The system includes an engine sound localizer configured to localize a first sound generated by the engine to cause a vehicle driver inside the cabin to hear the first sound from a first position, and an audio sound localizer configured to localize a second sound generated by the audio device to cause the driver to hear the second sound from a second position. The first and second positions are located forward of the driver in front-and-rear directions of the vehicle and separated from each other by a given distance.

Abstract: A panel loudspeaker controller for controlling a panel loudspeaker including a plurality of actuators, the panel loudspeaker controller including a plurality of electrical signal inputs, each input being associated with each actuator of the panel loudspeaker to be controlled; a plurality of signal processors, each signal processor being associated with each input and having an output for an electrical signal to control an actuator of the panel loudspeaker, and each signal processor implementing a transfer function from its input to its output based on each actuator of the panel loudspeaker to a desired acoustic receiver; and a signal processor controller associated with all of the plurality of signal processors, wherein the signal processor controller is preconfigured to improve phase alignment between the signals as an ensemble output at the outputs of the signal processors.