Abstract: An audio-enhanced furniture system comprises: a furniture assembly; an upholstery fabric at least partially covering the furniture assembly; and a speaker system positioned within the furniture assembly, the speaker system including a speaker covered by the upholstery fabric. The speaker is configured to be tuned to compensate for sound being emitted from the speaker through the upholstery fabric.

Abstract: Loudspeakers are described that may reduce comb filtering effects perceived by a listener by either 1) moving transducers closer to a sound reflective surface (e.g., a baseplate, a tabletop or a floor) through vertical (height) or rotational adjustments of the transducers or 2) guiding sound produced by the transducers to be released into the listening area proximate to the reflective surface through the use of horns and openings that are at a prescribed distance from the reflective surface. The reduction of this distance between the reflective surface and the point at which sound emitted by the transducers is released into the listening area may lead to shorter reflected path that reduces comb filtering effects caused by reflected sounds that are delayed relative to the direct sound. Accordingly, the loudspeakers shown and described may be placed on reflective surfaces without severe audio coloration caused by reflected sounds.

Abstract: A vehicle and a controlling method of the same include a speaker outputting virtual engine sound integrated with a head lamp. The vehicle includes the head lamp; a head lamp case in which the head lamp is provided; and a speaker for outputting a virtual engine sound, and provided inside the head lamp case, where an internal cross-sectional area of the head lamp case may increase in a direction toward a front of the vehicle based on a position where the speaker is provided.

Abstract: Example techniques facilitate calibration of a playback device. An example implementation involves a computing device capturing, via a microphone, data representing multiple iterations of a calibration sound as played by a playback device. The computing device identifies multiple sections within the captured data. Two or more sections represent respective iterations of the calibration sound as played by the playback device. Based on the multiple identified sections, the computing device determines a frequency response of the playback device, the frequency response of the playback device representing audio output by the playback device and acoustic characteristics of an environment around the playback device. Based on the frequency response of the playback device and a target frequency response, the computing device determines one or more parameters of an audio processing algorithm and sends, to the playback device, the one or more parameters of the audio processing algorithm.

Abstract: An example method of operation may include determining a frequency response to a measured chirp signal detected from one or more speakers, determining an average value of the frequency response based on a high limit value and a low limit value, subtracting a measured response from a target response, and the target response is based on one or more filter frequencies; determining a frequency limited target filter with audible parameters based on the subtraction, and applying an infinite impulse response (IIR) biquad filter based on an area defined by the frequency limited target filter to equalize the frequency response of the one or more speakers.

Abstract: There is provided a method for controlling bass reproduction properties of a multichannel audio system, wherein the audio system has inputs for at least two audio input signals and includes a set of loudspeakers, including at least one bass-capable loudspeaker and at least two high-range loudspeakers, each loudspeaker being associated with a loudspeaker channel. The method includes obtaining impulse responses or transfer functions that represent the sound reproduction properties of each loudspeaker channel at a number of measurement or control positions. The method also includes tuning, when the audio system includes more than one bass-capable loudspeaker, loudspeaker channels of at least two bass loudspeakers to each other so that their sum impulse response has minimum spatial variability, and/or controlling high-range loudspeaker speaker channels to be in-phase with each other and/or with bass-capable loudspeaker channel in a crossover frequency band.

Abstract: An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Systems and methods utilize a modified genetic algorithm for adapting an off-the-shelf audio system, such as in a high-end television, to a given, particular room or other physical location presenting a specific or unique auditory environment with a set of acoustic properties. An audio system is adapted to a given room by determining an IIR based EQ solution via iterative techniques, including an iterative technique based upon a genetic algorithm adapted for an audio frequency response equalization application. In a variant, an audio system is adapted to a particular room, adjust the EQ across a microphone's bandwidth while preserving the factory-calibrated EQ response across the remaining bandwidth.

Abstract: An electronic device includes a balloon configured to contact a surface of a human conduit and one or more a biometric sensors operatively coupled to or on or in or within the balloon for detecting a biometric signal. The electronic device in some examples is a biometric sensor or measuring device. In other examples, the electronic device is an integrated device such as an earpiece having biometric sensors. In yet other examples, the electronic device can be operatively coupled to other device and other biometric sensors. Other embodiments are disclosed.

Abstract: Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

Abstract: A radar sensor system is provided. The radar sensor system includes: at least two radar sensors each having at least one transmitter and at least one receiver, detection regions of the two radar sensors overlapping at least partially. The two radar sensors are situated at a defined distance from one another. Transmit signals of the two radar sensors are synchronizable in such a way that radiation of one radar sensor that was emitted by the respective other radar sensor and reflected by an object is capable of being evaluated by an evaluation device.

Abstract: The present invention discloses a method for producing parametric sound using parametric sound system which is based on ultrasonic electrostatic transducers. It comprises modulation of a carrier ultrasonic signal with a processed audio signal in audio signal processor comprising adaptive frequency filtering based on the audio signal level, dynamic range compression, square root operation, amplification of the modulated ultrasonic signal using a D-class amplifier, driving an electrostatic transducer and generating modulated ultrasonic waves into the air. The electrostatic transducer for the parametric sound system comprises a specific back plate structure that improves electromechanical efficiency of the transducer and also enables realization of a phased array on a single back plate. The disclosed manufacturing method of the electrostatic transducer comprises producing sets of electrodes on the surface of the back plate forming individual cells.

Abstract: An example method of operation may include identifying, in a particular room environment, a number of speakers and one or more microphones on a network controlled by a controller and amplifier, providing test signals to play sequentially from each amplifier channel of the amplifier and the speakers, monitoring the test signals from the one or more microphones simultaneously to detect operational speakers and amplifier channels, providing additional test signals to the speakers to determine tuning parameters, detecting the additional test signals at the one or more microphones controlled by the controller, and automatically establishing a background noise level and noise spectrum of the room environment based on the detected additional test signals.

Abstract: An example playback device is configured to receive a first stream of audio comprising source audio content to be played back by the playback device and record, via one or more microphones of the playback device, an audio signal output by the playback device based on the playback device playing the source audio content. The playback device is also configured to determine a transfer function between a frequency-domain representation of the first stream of audio and a frequency-domain representation of the recorded audio signal, and then determine an estimated frequency response of the playback device based on a difference between (i) the transfer function and (ii) a self-response of the playback device, where the self-response of the playback device is stored in a memory of the playback device. Based on the estimated frequency response, the playback device is configured to determine an acoustic calibration adjustment and implement the acoustic calibration adjustment.

Abstract: A system for detecting a presence of a feature or function of an output device used with an electronic device. The system includes a connector associated with the output device, the connector having a magnetic region comprising one or more magnetized members to provide a magnetic field. The system further includes a magnetic sensor associated with the electronic device, the magnetic sensor being positioned proximate a connection port of the electronic device, the connection port being configured to receive the connector to enable the magnetic sensor to detect a presence of the magnetic field and to determine the feature or function of the output device. The system further includes control logic associated with the magnetic sensor, the control logic controlling the electronic device based on the presence of the magnetic field and according to the determined feature or function of the output device.

Abstract: The present disclosure provides a method, an apparatus and a device for processing a sound signal, wherein the method comprises: acquiring a transmitted signal spectrum of a target sound signal sent out by a loudspeaker and a received signal spectrum of the target sound signal received by a microphone; detecting whether there is a signal distortion frequency band with signal distortion in the target sound signal according to the transmitted signal spectrum and the received signal spectrum, and when detecting that the signal distortion frequency band exists, performing compression processing on the target sound signal according to the signal distortion frequency band during a current signal processing cycle, and transmitting a compressed target sound signal through the loudspeaker.

Abstract: Examples described herein involve a playback device performing one or more playback device actions based on locations of one or more physical contacts on an external surface of the playback device. A processor of the playback device may receive from an array of proximity sensors underlying an external surface of the playback device, location data indicating a physical contact at a location on the external surface. Based on at least the location, the processor may identify a playback device action from a plurality of playback device actions, and cause at least the playback device to perform the identified playback device action. The playback device may further include at least one orientation sensor from which the processor may also receive orientation data indicating an orientation of the playback device. The processor may identify the playback device action also based on the orientation of the playback device.

Abstract: An audio processor for providing loudspeaker signals on the basis of input signals, like channel signals and/or object signals, obtains an information about the position of a listener and an information about the position of a plurality of loudspeakers, or sound transducers. The audio processor selects one or more loudspeakers for a rendering of the objects and/or of the channel objects and/or of the adapted signals, derived from the input signals. The selection depends on the information about the position of the listener and about the positions of the loudspeakers and takes into consideration the information about one or more acoustic obstacles. The audio signal processor renders the objects/channel objects/adapted signals derived from the input signals, in dependence on the information about the position of the listener and about positions of the loudspeakers, in order to obtain the loudspeaker signals, such that a rendered sound follows a listener.

Abstract: A 3D sound spatializer provides delay-compensated HRTF interpolation techniques and efficient cross-fading between current and delayed HRTF filter results to mitigate artifacts caused by interpolation between HRTF filters and the use of time-varying HRTF filters.

Abstract: A sound control device controls an interior sound in a vehicle based on a slope of a road on which the vehicle travels. The sound control device includes: a sound canceling circuit for generating a first correcting sound for lowering a level of an interior sound measured in the vehicle; a sound boosting circuit for generating a second correcting sound for increasing the level of the interior sound; and a controller for setting a level of a second target sound to be less than a level of a first target sound. In particular, the second target corresponds to a ramp in which an absolute value of the slope is greater than a reference value and the first target sound corresponds to a flatland in which the absolute value of the slope is equal to or less than the reference value.

Abstract: Systems and methods are disclosed in which a playback device transmits a first sound signal including a predetermined waveform. In one example, the playback device receives a second sound signal including at least one reflection of the first sound signal. The second sound signal is processed to determine a location of a person relative to the playback device, and a characteristic of audio reproduction by the playback device is selected, based on the determined location of the person.

Abstract: Example techniques may involve headphone interaction. An implementation may include while headphones are disconnected from a control device, the control device receiving an indication of particular audio content being played back by a first zone of the media playback system. While the particular audio content is being played back by the one or more playback devices of the media playback system, detecting that headphones have been connected to the first control device. Headphones may be connectable to the control device via either (a) an analog headphone jack or (b) a point-to-point personal area network connection. In response to detecting that the headphones are connected to the control device: the control device (i) causing the one or more playback devices of the first zone to stop playback of the particular audio content; (ii) retrieving the particular audio content and (iii) resuming playback of the particular audio content via the connected headphones.

Abstract: An apparatus including circuitry configured for: obtaining at least one spatial audio signal including at least one audio signal, wherein the at least one spatial audio signal defines an audio scene forming at least in part an immersive media content; obtaining at least one augmentation control parameter associated with the spatial audio signal, wherein the at least one augmentation control parameter is configured to control at least in part a rendering of the audio scene; and transmitting/storing the at least one spatial audio signals and the at least one augmentation control parameter, the at least one spatial audio signal and the at least one augmentation control parameter being received/retrieved at a renderer so as to control at least in part rendering of the audio scene based on the at least one augmentation control parameter.

Abstract: An example implementation involves a computing device transmitting, via a local area network, a command that instructs a playback device to play a particular audio signal. The example implementation also involves the computing device receiving data indicating a detected audio signal corresponding to playback of the particular audio signal by the playback device, where the detected audio signal includes a portion of the particular audio signal. The implementation further involves the computing device obtaining data indicating a predetermined audio characteristic and determining an audio processing algorithm based on the detected audio signal and the predetermined audio characteristic. The example implementation involves causing the playback device to apply the determined audio processing algorithm when playing audio via at least one speaker.

Abstract: A monitoring system for locating and classifying an event in a monitoring area by a computation unit including a visual 3D capturing unit providing geometric 3D information and an acoustic capturing unit providing an acoustic information of the monitoring area. An event detector is configured with an acoustic channel and a visual channel to detect the event. The acoustic channel is configured to detect the event as a sound event in the acoustic information and to determine a localization of the sound. The visual channel is configured to detect the event as a visual event in the geometric 3D information and to derive a localization of the visual event. The event detector provides detected events with a region of interest for detected event, which is analyzed in order to assign the detected event a class within a plurality of event classes.

Abstract: A parametric equalizer includes an equalizer circuit, a first protection circuit, a second protection circuit, and a first addition circuit. The equalizer circuit is arranged to receive an input signal, and process the input signal to generate an output signal. The first protection circuit is arranged to generate a first protection signal according to the output signal, the input signal, and a first processed signal. The second protection circuit is arranged to generate a second protection signal according to the input signal and a second processed signal. The first addition circuit is coupled to the first protection circuit and the second protection circuit, and is arranged to combine the first protection signal and the second protection signal to generate an equalizer output signal.

Abstract: A sound signal processing device that acquires a collected-sound signal obtained by sampling a sound collected by a microphone at a first sampling frequency, and receives a reproduced-sound signal obtained by sampling a sound for reproduction at a second sampling frequency different from the first sampling frequency, and then converts the sampling frequency of the reproduced-sound signal into the first sampling frequency so as to remove an acoustic echo from the collected-sound signal by using the reproduced-sound signal whose sampling frequency has been converted.

Abstract: Frequency domain compensation is provided for spectral impairment resulting from the audio path characteristics for a given audio device in a given listening space. Selected segments of an audio stream are recorded at a listener position to measure degradation in the audio path and to update compensation filter characteristics of the audio device. Recorded transmitted and received audio sequences are aligned based and compared in the frequency domain. The difference between the aligned transmitted and received sequences represents the frequency domain degradation along the acoustic path due to the speaker, the physical attributes of the room, and noise. A dynamically updated noise model is determined for adjusting compensation filter characteristics of the audio device, which can be updated during use of the audio device. A compensation curve is derived which can adapt the equalization of the audio device passively during normal usage.

Abstract: A method for estimating an acoustic influence of walls of a room, comprising emitting a known excitation sound signal, receiving a set of measurement signals, each measurement signal being received by one microphone in a microphone array and each measurement signal including a set of echoes caused by reflections by the walls, solving a linear system of equations to identify locations of image source and estimating the acoustic influence based these image sources. The signal model includes a convolution of: the excitation signal, a multichannel filter (M) representing the relative delays of the microphones in the microphone array, the relative delays determined based on a known geometry of the microphone array, and a directivity model ?(n, p) of the driver(s) in the form of an anechoic far-field impulse response as a function of transmit angle.

Abstract: A method for adjusting sound playback of a portable device for constancy notwithstanding different environments outputs from the portable device detectable audio signals inaudible to user and the device receives reflected audio before the portable device is actually commanded to play an audio file. A list of volume weightings for reflected audio is calculated. Before commencing playback of the audio file, the portable device obtains reference volume weightings from a list according to the current volume setting, and calculates adjustment coefficients for different frequency bands based on weightings of the reference volume list and of the reflected audio list. The audio signals of the audio file are output after adjustment. A portable device is also disclosed.

Abstract: A thermal excitation acoustic-wave-generating device includes a first heating element, a substrate that includes a main surface along which the first heating element is disposed, and a facing body that faces the substrate with the first heating element interposed therebetween. The substrate and the facing body define a path for an acoustic wave. A length of the path is close to a whole number multiple of ¼ of a wavelength of the acoustic wave.

Abstract: Embodiments of the disclosure provide methods and apparatuses processing audio data. The method can include: acquiring audio data by an audio capturing device, determining feature information of an enclosure in which the audio capturing device is located, and reverberating the feature information into the audio data.

Abstract: The application describes techniques for adaptively setting a threshold that will be used to determine the gain applied to a haptics signal.

Abstract: Example techniques involve suggesting an application of a streaming audio service via an application of a media playback system. An example implementation involves a mobile device displaying, via a control application of a media playback system, a library control comprising indicators representing respective media items of a particular streaming audio service. The mobile device receives, via the library control, input data indicating a selection of an indicator representing a particular media item. The mobile device sends instructions to cause a playback device to play back the particular media item. While the playback device is playing back the particular media item, the mobile device displays, via the control application, a control representing an application of the particular streaming audio service. The mobile device receives input data indicating a selection of the control representing the application of the particular streaming audio service and causes the mobile device to download the application.

Abstract: The present disclosure relates to a device for playing an acoustic sound and a touch sensation, and more particularly, to a device for emitting a sound to a front surface of a speaker and playing a touch sensation feedback to a rear surface of the speaker through a driving modulation using two or more speakers while simultaneously concentrating an acoustic pressure generated according to a vibration of a membrane of the speaker to a surface of a module through a movement pipe when an audio signal is applied.

Abstract: A method of equalising an audio signal derived from a microphone, the method comprising: receiving the audio signal; applying an order-statistic filter to the audio signal in the frequency domain to generate a statistically filtered audio signal; equalising the received audio signal based on the statistically filtered audio signal to generate an equalised audio signal.

Abstract: Waveform-coding is applied to map successive on-off-keying (OOK) data bits onto successive multicarrier modulated symbols in time domain, wherein each multicarrier modulated symbol includes a set of sub-carriers in which alternating sub-carriers are set to non-zeros and zeros in frequency domain. The waveform coded multicarrier modulated symbols are up-converted to a carrier frequency to provide a data signal that is transmitted over a wireless channel.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Certain embodiments provide methods and systems for managing a sound profile. An example playback device includes a network interface and a non-transitory computer readable storage medium having stored therein instructions executable by the processor. When executed by the processor, the instructions are to configure the playback device to receive, via the network interface over a local area network (LAN) from a controller device, an instruction. The example playback device is to obtain, based on the instruction, via the network interface from a location outside of the LAN, data comprising a sound profile. The example playback device is to update one or more parameters at the playback device based on the sound profile. The example playback device is to play back an audio signal according to the sound profile.

Abstract: An electronic apparatus and method for configuration of an audio reproduction system is provided. The electronic apparatus receives image of a listening environment and applies a machine learning model on the image to identify a plurality of objects, including a display device and a plurality of audio devices. The electronic apparatus determines contour information of each identified object. The electronic apparatus retrieves real-dimension information of each identified object. Based on the determined contour information and the retrieved real-dimension information, the electronic apparatus determines first distance information between a listening position and each identified object. The electronic apparatus receives an audio signal from each audio device and determines a second distance between each audio device and the listening position based on the received audio signal.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to generate equalization adjustments for an audio signal includes an input feature set generator to generate an equalization input feature set by generating a data structure corresponding to a user input, the data structure including a number of entries identifying a selected variant of music and including the data structure in the equalization input feature set, and a model executor to, in response to obtaining the equalization input feature set, adjust at least one weight of a neural network model to generate the equalization adjustments for the audio signal.

Abstract: A vehicle includes a first voice output section, a second voice output section, a third voice output section, and a control section which are arranged side by side in a width direction of the vehicle. The control section controls volume of the voice output from the first, second and third voice output sections. If sitting on a rider seat and a pillion passenger seat is detected, the control section controls the first, second and third voice output sections to make the volume of the voice output from the first and third voice output sections larger than the volume of the voice output from the second voice output section. The first and third voice output sections are located at both ends in the width direction of the vehicle and the second voice output section is located between the first and third voice output sections.

Abstract: A method for adjusting sound playback of a portable device for constancy notwithstanding different environments outputs from the portable device detectable audio signals inaudible to user and the device receives reflected audio before the portable device is actually commanded to play an audio file. A list of volume weightings for reflected audio is calculated. Before commencing playback of the audio file, the portable device obtains reference volume weightings from a list according to the current volume setting, and calculates adjustment coefficients for different frequency bands based on weightings of the reference volume list and of the reflected audio list. The audio signals of the audio file are output after adjustment. A portable device is also disclosed.

Abstract: The invention discloses a prediction method for porous material of electroacoustic devices and prediction system thereof. The method comprises the following steps. The step (A) is to obtain at least one acoustic parameter of a porous material from an electroacoustic device, and the at least one acoustic parameter comprises a flow resistance value, a specific flow resistance value and a flow resistance ratio. The step (B) is to calculate an actual resistance value of the porous material based on the at least one acoustic parameter. Thereafter, the step (C) establishes an equivalent circuit model corresponding to the electroacoustic device based on the structure configuration and material parameters of the electroacoustic device. At last, step (D) introduces the actual impedance value of the porous material into the equivalent circuit model, and calculates the frequency response curve and impedance curve of the electroacoustic device affected by the porous material.

Abstract: A mixing device of a first signal and a second signal on a time-frequency plane, includes a control signal generation unit to generate a control signal indicating whether to perform prioritized mixing that includes amplification of the first signal and attenuation of the second signal; and a gain derivation unit to derive a first gain for amplifying the first signal and a second gain for attenuating the second signal based on the control signal. The control signal takes at least a first value and a second value different from the first value, wherein the first value is not continuous beyond a predetermined bandwidth on a frequency axis. The mixing device applies the prioritized mixing to the first and second signals when the control signal indicates the first value, and applies simple addition to the first and second signals when the control signal indicates the second value.

Abstract: The present application relates to a loudness adjustment method and apparatus, and an electronic device and a storage medium, mainly relating to the technical field of multimedia. The method comprises: converting a sound signal of a multimedia resource into a first frequency domain signal; acquiring a second frequency domain signal based on the first frequency domain signal and frequency response information of a current electronic device, wherein the second frequency domain signal is used for reflecting the loudness of the first frequency domain signal when same is played on the current electronic device; and adjusting the loudness of the sound signal based on the second frequency domain signal and a target loudness, so as to obtain a target sound signal of the multimedia resource.

Abstract: A microphone capsule assembly includes a first microphone capsule oriented to face a first direction that is parallel to a first plane, a second microphone capsule oriented to face in a direction opposite to the first direction, where a front face of the second microphone capsule is spaced a distance from a front face of the first microphone capsule in the first direction, a third microphone capsule oriented to face in a second direction that is at a first acute angle to the first direction when measured parallel to the first plane, and a fourth microphone capsule oriented to face in a third direction that is at a second acute angle to the first direction when measured parallel to the first plane, where the third and fourth microphone capsules are spaced a distance from the front face of the first microphone capsule in the first direction.

Abstract: A hearing aid system is provided that facilitates adjustment of signal processing parameters ? of the hearing aid system with minimum user intervention, wherein the hearing aid system is capable of calculating signal processing parameters ? for evaluation of the user when the user has entered an input, e.g. using a smartwatch, to this effect. The evaluation takes place for a certain time period and in the event that the user has entered a consent input indicating that he or she is pleased with the set ? of signal processing parameters under evaluation, the hearing aid system continues processing with those signal processing parameters; and if the user is not pleased with the signal processing parameters ? under evaluation, the hearing aid system calculates another set {circumflex over (?)} of signal processing parameters for evaluation of the user.

Abstract: A mixing apparatus comprises a weighting circuit configured to, within a particular control target band, perform weighting amplitude spectra of a priority audio signal and a non-priority audio signal taking frequency masking and time masking into account, an amplitude changer configured to relatively amplify an amplitude spectrum of the priority audio signal within the control target band based on the amplitude spectrum to which the weighting has been applied, a phase changer configured to make a phase spectrum of the non-priority audio signal approach a phase spectrum of the priority audio signal, within the control target band, based on the amplitude spectrum to which the weighting has been applied, and a mixer configured to mix the priority audio signal and the non-priority audio signal after the amplitude spectrum and the phase spectrum are changed.