Abstract: Systems and methods for enhancing a headset user's own voice include at least two outside microphones, an inside microphone, audio input components operable to receive and process the microphone signals, a voice activity detector operable to detect speech presence and absence in the received and/or processed signals, and a cross-over module configured to generate an enhanced voice signal. The audio processing components includes a low frequency branch comprising low pass filter banks, a low frequency spatial filter, a low frequency spectral filter and an equalizer, and a high frequency branch comprising highpass filter banks, a high frequency spatial filter, and a high frequency spectral filter.

Abstract: A signal processing system and method for delivering spatialized sound by optimizing sound waveforms from a sparse array of speakers to the ears of a user. The system can provide listening areas within a room or space, to provide spatialization sounds to create a 3D audio effect. In a binaural mode, a binary speaker array provides targeted beams aimed towards a user's ears.

Abstract: An apparatus comprises a receiver (201) receiving a multi-channel audio signal representing audio for a scene. An extractor (203) extracts at least one directional audio component by applying a spatial filtering to the multi-channel signal where the spatial filtering is dependent on the multi-channel audio signal. A feature processor (205) determines a set of features for the first directional audio component and a categorizer (207) determines a first audio source category out of a plurality of audio source categories for the directional audio signal in response to the set of features. An assigner (209) assigns a first audio source property to the first directional audio component from a set of audio source properties for the first audio source category. The apparatus may provide very advantageous categorization and characterization of individual audio sources/components present in a multi-channel signal. This may be advantageous e.g. for visualization of audio events.

Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).

Abstract: An example may include receiving a page code identifier, determining a priority of the page code identifier, queuing the page code identifier in a paging queue, retrieving content associated with the page code, and forwarding the content to one or more audio devices identified by the page code identifier when the page code has reached a top of the queue.

Abstract: According to some embodiments, a control device for a premises security system that is configured to monitor a premises using a plurality of premises devices is provided. The control device is configured to obtain a plurality of acoustic samples for the premises, detect a sound anomaly in at least one of the plurality of acoustic samples, obtain a verification that the sound anomaly is expected, generate an acoustic model for the premises based at least on the plurality of acoustic samples and the verification that the sound anomaly is expected, receive data representing a detected sound during monitoring of the premises, compare the detected sound with the acoustic model for the premises to determine that the detected sound is unexpected, and initiate a premises security system alert based at least on the detected sound being unexpected.

Abstract: Example embodiments disclosed herein relate to audio object clustering. A method for metadata-preserved audio object clustering is disclosed. The method comprises classifying an audio object into at least a category based rendering mode information metadata. The method further comprises assigning a predetermined number of clusters to the categories and rendering the audio object based on the rendering mode. Corresponding system and computer program product are also disclosed.

Abstract: Disclosed are a sound tracing apparatus and a sound tracing method, and the sound tracing apparatus includes a first acceleration structure generation unit configured to generate a first acceleration structure for a static scene in a sound space, an intersection test execution unit configured to perform an intersection test on each of a plurality of dynamic objects constituting a dynamic scene in the sound space to detect whether or not the dynamic object affects a sound propagation path, a second acceleration structure generation unit configured to select the dynamic objects that affect the sound propagation path as a result of the intersection test and then generate the second acceleration structure for the dynamic scene, and a sound generation unit configured to generate a 3D sound by performing sound tracing based on the first and second acceleration structures.

Abstract: An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

Abstract: Methods, systems, and apparatus, for a wearable aural development medical device and system. A medical device is worn by an infant while the infant is in a neonatal intensive care unit (NICU). In particular, the device protects the child's hearing by active filtering, while, at the same time, provides aural stimulation for development. In some implementations, a remote server system also enables people associated the child, e.g., parents and siblings, to provide recordings, such as recordings of their voices, for playback to the child.

Abstract: A system for sound localization can include a first electronic device having a microphone to detect a sound, and a second electronic device. A processor can be in communication with the first electronic device and the second electronic device. The processor can receive a first signal from the first electronic device corresponding to the detected sound, determine a location of origin of the detected sound based at least in part on the first signal, and provide a second signal to the second electronic device based at least in part on the location of origin.

Abstract: Disclosed herein, among other things, are methods and apparatus for providing a time-stamp based controller for synchronization of sink or source sampling rate with external packet rate. A method for wireless communications includes receiving a transmission of a packet using a wireless transceiver of an electronic device, and using a processor of the electronic device to read a first value of a system timer and store the first value as an arrival time-stamp. The packet is decoded and processed by the processor, and sent to an output. When the processed packet is sent, a second value of the system timer is read, adjusted and stored as a departure time-stamp. The arrival time-stamp and the departure time-stamp are used to calculate an adjustment stimulus for a sample rate actuator of the electronic device. The sample rate actuator is configured to maintain synchronization of sampling rate with an external packet rate.

Abstract: An audio processing method includes processing, by M first virtual speakers, a to-be-processed audio signal to obtain M first audio signals; processing, by N second virtual speakers, the to-be-processed audio signal to obtain N second audio signals; obtain M first head-related transfer functions (HRTFs) centered at a left ear position and N second HRTFs centered at a right ear position; obtain a first target audio signal based on the M first audio signals and the M first HRTFs; and obtain a second target audio signal based on the N second audio signals and the N second HRTFs.

Abstract: A signal processing device for processing audio signals is described. The signal processing device has an input interface for receiving an input signal and an output interface for outputting an output signal. Moreover, the signal processing device has at least one first neural network for conditioning the input signal and at least one second neural network for separating one or more audio signals from the input signal. The at least one first neural network and the at least one second neural network are arranged sequentially.

Abstract: An apparatus for generating loudspeaker signals includes an object metadata processor configured to receive metadata, to calculate a second position of the audio object depending on the first position of the audio object and on a size of a screen if the audio object is indicated in the metadata as being screen-related, to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, and to feed the second position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being screen-related. The apparatus further includes an object renderer configured to receive an audio object and to generate the loudspeaker signals depending on the audio object and on position information.

Abstract: A method for steering binauralization of audio is provided. The method comprises steps of: receiving (410) an audio input signal, calculating (430) a confidence value indicating a likelihood that a current audio frame of the audio input signal comprises binauralized audio; determining (450) a state signal based on the confidence value; determining (460) a steering signal, based on the first confidence value, the state signal and an energy value of the audio frame; and generating (470) an audio output signal with steered binauralization by processing the audio input signal according to the steering signal.

Abstract: A sound signal processing method includes: obtaining a sound signal; obtaining impulse response data that was measured in a predetermined space before the sound signal is obtained; generating an early reflected sound control signal not including a reverberant sound by convolving impulse response data of an early reflected sound among the obtained impulse response data into the obtained sound signal.

Abstract: Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

Abstract: An electronic device according to an embodiment disclosed herein may include a sensor, a short-range communication module, and a processor. The processor is configured to generate first rotation angle information, establish a connection to the external electronic device, receive, from the external electronic device, second rotation angle information and check data uniquely assigned to the second rotation angle information, configure a first time stamp based on the check data and a time of receiving the second rotation angle information, configure a second time stamp based on the check data and a time of correcting the second rotation angle information, compare the first time stamp and the second time stamp to calculate a delay time, correct the second rotation angle information based on the delay time, and generate posture information based on the first rotation angle information and the corrected second rotation angle information.

Abstract: A decoder for generating an audio output signal having one or more audio output channels is provided, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal. Moreover, an encoder is provided.

Abstract: Provided is a sound image localizing device, a sound image localizing method, and a program that enable a virtual speaker to reproduce sound in a wide frequency band with high sound quality. A sound image localizing device 10 includes a directivity control filter design unit 11 that computes a directivity control filter from a desired directional characteristic, a filter coefficient correction unit 12 that corrects the directivity control filter computed by the directivity control filter design unit 11, and a convolution operation unit 13 that computes an output acoustic signal by performing convolution of an input acoustic signal and the directivity control filter corrected by the filter coefficient correction unit 12.

Abstract: A multifunctional earphone system for sports activities is described which comprises the following: a first apparatus configured to be carried in one of a user's ears, the first apparatus comprising a first data communication unit and a first loudspeaker, and a second apparatus configured to be carried in the user's other ear, the second apparatus comprising a second data communication unit and a second loudspeaker, wherein at least one of the first apparatus and the second apparatus comprises a sensor unit and a data processing unit, wherein the data processing unit is configured to generate performance data based on measurement data acquired by the sensor unit, wherein the first apparatus further comprises a signal processing unit configured to generate a binaural audio signal based on the performance data, the binaural audio signal comprising a first signal part to be output by the first loudspeaker and a second signal part to be output by the second loudspeaker, and wherein the first data communication un

Abstract: A method for acoustic damping of sound clips includes identifying an audio clip for a location of a user in an environment and fragmenting the audio clip into a plurality of sound clips. The method further includes, responsive to determining at least one sound clip from the audio clip requires acoustic damping, performing the acoustic damping on the at least one sound clip, where a damping ratio for the at least one sound clip is altered. The method further includes responsive to determining to stitch the plurality of sound clips, stitching the plurality of sounds clips to form the audio clip, where the plurality of sound clips includes the at least one sound clip with the acoustic damping. The method further includes displaying a visual representation of the audio clip with the plurality of sound clips.

Abstract: Methods and apparatus assist listeners in distinguishing between electronically generated binaural sound and physical environment sound while the listener wears a wearable electronic device that provides the binaural sound to the listener. The wearable electronic device generates a visual alert or audio alert when the electronically generated binaural sound occurs.

Abstract: To provide a realistic video and audio to a passenger of a moving body, an output control device includes: a position and orientation identifying section which identifies a real position and a real orientation; a reading section which reads virtual space data; a virtual position determining section which determines a virtual position and a real orientation according to the real position and the real orientation; a display control section which controls a display to display, based on the virtual position, the virtual orientation, and a position where the display (4) is provided, an image of a partial region or a part of an object in a virtual space; and an audio output control section which controls, based on the virtual orientation and a positional relationship between the virtual position and an audio source object, a position of a stereo image formed by audio outputted from a speaker.

Abstract: An audio rendering system includes a processor that combines audio input signals with personalized spatial audio transfer functions preferably including room responses. The personalized spatial audio transfer functions are selected from a database having a plurality of candidate transfer function datasets derived from in-ear microphone measurements for a plurality of individuals. Alternatively, the personalized transfer function datasets are derived from actual in-ear measurements of the listener. Foreground and background positions are designated and matched with transfer function pairs from the selected dataset for the foreground and background direction and distance. Two channels of input audio such as voice and music are processed. When a voice communication such as a phone call is accepted the music being rendered is moved from a foreground to a background channel corresponding to a background spatial audio position using the personalized transfer functions.

Abstract: An apparatus configured to: determine, for two or more microphone audio signals, a plurality of spatial audio parameters for providing spatial audio reproduction, wherein the plurality of spatial audio parameters are associated with respective frequency bands of at least two frequency bands of the two or more microphone audio signals; determine at least one coherence parameter associated with a sound field, wherein the sound field is associated with the two or more microphone audio signals; determine at least one audio signal based on the two or more microphone audio signals; and enable the spatial audio reproduction based on the plurality of spatial audio parameters, the at least one coherence parameter, and the at least one determined audio signal.

Abstract: An apparatus configured to: determine, for two or more speaker channel audio signals, at least one spatial audio parameter for providing spatial audio reproduction; determine, between the two or more speaker channel audio signals, at least one coherence parameter, wherein the at least one coherence parameter is configured to provide at least one inter-channel coherence information between the two or more speaker channel audio signals for respective ones of at least two frequency bands of the two or more speaker channel audio signals; determine at least one value based, at least partially, on the at least one coherence information, wherein the at least one value is configured to indicate at least one information associated with the at least one inter-channel coherence information; and transmit the at least one spatial audio parameter and the at least one determined value.

Abstract: Embodiments relate to binaural spatialization of more than two sound sources on two audio channels of an audio system. Sound signals each emitted from a corresponding sound source are collected, and a respective virtual position within an angular range of a sound scene is assigned to each sound source. Multi-source audio signals are generated by panning each sound signal according to the respective virtual position. A first multi-source audio signal is spatialized to a first direction to generate a first left signal and a first right signal. A second multi-source audio signal is spatialized to a second direction to generate a second left signal and a second right signal. A binaural signal is generated using the first left signal, the second left signal, the first right signal, and the second right signal. The binaural signal is such that each sound source appears to originate from its respective virtual position.

Abstract: A first subwoofer may be configured to output multimedia content in synchrony with at least one other playback device and a second subwoofer. The first subwoofer may, based on a received indication of an acoustic characteristic of the at least one other playback device, determine a crossover frequency of (i) the first subwoofer and the second subwoofer and (ii) the at least one other playback device. After determining the crossover frequency, the first subwoofer may output a first tone set and a second tone set in synchrony with the second subwoofer and the at least one other playback device, and after outputting the first tone set and the second tone set, receive, from a controller device, an indication of a selected one of the first tone set or the second tone set. Based on the selected tone set, the first subwoofer may adjust a phase setting of the first subwoofer.

Abstract: An electronic device may include control circuitry that allows a user to listen to audio playing from an external speaker in the environment from the user's personal speaker device such as a pair of headphones, earbuds, or a personal loudspeaker. In response to receiving user input, the control circuitry may gather audio input from the environment with a microphone. The control circuitry may process the detected audio input to identify what audio track is currently playing from the external speaker and to determine which part of the audio track is currently playing. The control circuitry may then start playing the identified audio track from the user's personal speaker device in sync with the audio track playing from the external speaker. The electronic device may search music streaming platforms, public radio station information, and other public music content sources to anticipate and buffer upcoming songs.

Abstract: A measurement device according to an embodiment includes: a base portion (10), a plurality of pillars (20, 30, 40) in an arc shape not directly facing one another having one end of each connected to the base portion and the other end of each coupled by a coupling portion (60), and a plurality of speakers (70) provided on each of the pillars and having distances to a certain location substantially uniform.

Abstract: In general, techniques are described for obtaining audio rendering information from a bitstream. A method of rendering audio data includes receiving, at an interface of a device, an encoded audio bitstream, storing, to a memory of the device, encoded audio data of the encoded audio bitstream, parsing, by one or more processors of the device, a portion of the encoded audio data stored to the memory to select a renderer for the encoded audio data, the selected renderer comprising one of an object-based renderer or an ambisonic renderer, rendering, by the one or more processors of the device, the encoded audio data using the selected renderer to generate one or more rendered speaker feeds, and outputting, by one or more loudspeakers of the device, the one or more rendered speaker feeds.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for playback using pre-processed profile information and personalization. Example apparatus disclosed herein include a synchronizer to, in response to receiving a media signal to be played on a playback device, access an equalization (EQ) profile corresponding to the media signal; an EQ personalization manager to generate a personalized EQ setting; and an EQ adjustment implementor to modify playback of the media signal on the playback device based on a blended equalization generated based on the EQ profile and the personalized EQ setting.

Abstract: Examples disclosed herein include a speaker. The speaker may include a group of microphones and a processor. The processor may determine a first speaker-channel identifier for a multi-speaker system at least partially responsive to a first tone captured at the group of microphones. The processor may also determine a position of a source of the captured first tone relative to the speaker at least partially responsive to position information derived from the captured first tone. The processor may also determine a second speaker-channel identifier at least partially responsive to the first speaker-channel identifier and the position of the source of the captured first tone. The processor may also determine speaker settings at least partially responsive to the second speaker-channel identifier. Related devices, systems and methods are also disclosed.

Abstract: A method for spatial audio signal processing, including determining, for two or more playback audio signals, at least one spatial audio parameter for providing spatial audio reproduction; determining between the two or more playback audio signals at least one audio signal relationship parameter, the at least one audio signal relationship parameter being associated with a determination of inter-channel signal relationship information between the two or more playback audio signals and for at least two frequency bands, such that the two or more playback audio signals are configured to be reproduced based on the at least one spatial audio parameter and the at least one audio signal relationship parameter.

Abstract: Media content playback is based on an identified geographic location of a sound source. A geolocation of a target venue is determined. A first geolocation of a media playback device is determined. An indication that the media playback device is within a threshold distance of the target venue is received. Media content associated with the target venue is transferred to the media playback device, wherein one or more characteristics of the media content are dynamically modified to simulate audio originating from the target device. The one or more characteristics of the media content can include audio content characteristics associated with a timing filter, a loudness filter, and a timbre filter.

Abstract: An apparatus and method for pitch-shifting an audio signal with low complexity are disclosed. The method includes identifying a distance between an audio object included in the audio signal and a listener, checking whether the distance between the audio object and the listener decreases, and performing stepwise stretching pitch-shifting of repeatedly using at least one of frequency components of the audio signal when the distance between the audio object and the listener decreases.

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: A system (20) includes a plurality of microphones (22), configured to generate different respective signals in response to acoustic waves (36) arriving at the microphones, and a processor (34). The processor is configured to receive the signals, to combine the signals into multiple channels, which correspond to different respective directions relative to the microphones by virtue of each channel representing any portion of the acoustic waves arriving from the corresponding direction with greater weight, relative to others of the directions, to calculate respective energy measures of the channels, to select one of the directions, in response to the energy measure for the channel corresponding to the selected direction passing one or more energy thresholds, and to output a combined signal representing the selected direction with greater weight, relative to others of the directions. Other embodiments are also described.

Abstract: An apparatus includes at least one memory, instructions, and processor circuitry to execute the instructions to track movement of a head of a user wearing earphones, the earphones to move with the movement of the head of the user, the earphones to be communicatively coupled to a computing device. The processor circuitry is to obtain media content, the media content including first audio data for a first channel and second audio data for a second channel. The processor circuitry is to adjust, based on the movement of the head of the user, the first audio data for the first channel and the second audio data for the second channel. The processor circuitry is to cause the adjusted first audio data and the adjusted second audio data to be played by the earphones.

Abstract: A correction method of acoustic characteristics of a plurality of speakers installed in a room, the correction method groups speakers, among the plurality of speakers, having identical attachment condition to the above-mentioned room, measures each of output characteristics of the grouped speakers, calculates an average characteristic of the output characteristics of the grouped speakers, and corrects each of the output characteristics of the grouped speakers based on the average characteristic.

Abstract: An audio mixer includes a user interface, panners, a first adder, a localization device, a second adder, and an output circuit. The user interface supplies a first parameter and a second parameter for each channel based on a user operation. The first parameter indicates a position in a right-left direction. The second parameter specifies internalization or externalization. The panners respectively correspond to channels and, based on the first parameter, pan a sound signal corresponding to the each channel to generate first stereo signals. The first adder generates a second stereo signal by mixing first stereo signals respectively corresponding to externalization channels. The localization device generates two third stereo signals. The second adder generates a fourth stereo signal by mixing the two third stereo signals and first stereo signals respectively corresponding to internalization channels. The output circuit outputs the fourth stereo signal.

Abstract: Disclosed is an audio signal processing method including: receiving a stereo signal; transforming the stereo signal into a frequency-domain signal; rendering the first signal based on a first ipsilateral filter coefficient; generating a frontal ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a second ipsilateral filter coefficient; generating a side ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a contralateral filter coefficient; generating a side contralateral signal relating to the frequency-domain signal; transforming an ipsilateral signal, generated by mixing the frontal ipsilateral signal and the side ipsilateral signal, and the side contralateral signal into a time-domain ipsilateral signal and a time-domain contralateral signal, which are time-domain signals, respectively; and generating a binaural signal by mixing the time-domain ipsilateral signal and the time-domain contralateral signal.

Abstract: A method (910) for rendering an audio signal in a virtual reality rendering environment (180) is described. The method (910) comprises rendering (911) an origin audio signal of an audio source (311, 312, 313) from an origin source position on an origin sphere (114) around an origin listening position (301) of a listener (181). Furthermore, the method (900) comprises determining (912) that the listener (181) moves from the origin listening position (301) to a destination listening position (302). In addition, the method (900) comprises determining (913) a destination source position of the audio source (311, 312, 313) on a destination sphere (114) around the destination listening position (302) based on the origin source position, and determining (914) a destination audio signal of the audio source (311, 312, 313) based on the origin audio signal.

Abstract: An example device includes a memory configured to store audio data and location data associated with a plurality of audio streams and one or more processors coupled to the memory. The one or more processors are configured to obtain a first location of a first audio stream that includes an audio source and obtain a second location of a second audio stream that includes the audio source. The one or more processors are configured to generate direction vectors originating at the first location and the second location, based on a location of the audio source and the first location, and the location of the audio source and the second location, respectively. The one or more processors are also configured to determine parameters that describe a vector field based on the first direction vector and the second direction vector.

Abstract: An apparatus for generating a description of a combined audio scene, includes: an input interface for receiving a first description of a first scene in a first format and a second description of a second scene in a second format, wherein the second format is different from the first format; a format converter for converting the first description into a common format and for converting the second description into the common format, when the second format is different from the common format; and a format combiner for combining the first description in the common format and the second description in the common format to obtain the combined audio scene.

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: Techniques for determining positions of devices within an environment are described herein. In some instances, an environment, such as a home or office of a user, may include an array of devices, some or all of which may couple to a network or to other devices via short-range wireless connections (e.g., Bluetooth®, Zigbee®, etc.). These devices may capture an array of data for providing to a central service, which is configured to analyze the data and, based on this analysis, determine a location of the devices relative to one another. That is, the central service may analyze the data to determine relative distances and orientations between the identified devices within the environment.

Abstract: An audio communication device includes: a sound position determiner that determines sound localization positions for N audio signals in a virtual space having first and second walls; N sound localizers each performing sound localization processing to localize sound in the sound localization position determined by the sound position determiner, and outputting localized sound signals; an adder that sums the N localized sound signals, and outputs a summed localized sound signal. Each sound localizer performs the processing using: a first head-related transfer function (HRTF) assuming that a sound wave emitted from the sound localization position of the sound localizer determined by the sound position determiner directly reaches each ear of a hearer virtually present at the hearer position; and a second HRTF assuming that the sound wave emitted from the sound localization position reaches each ear of the hearer after being reflected by closer one of the first and second walls.