Abstract: Examples described herein involve identifying one or more error conditions during calibration of one or more playback devices in a playback environment. A microphone of a network device may detect and sample an audio signal while the one or more playback devices in the playback environment plays a calibration tone. A processor of the network device may then receive, from the microphone, a stream of audio data. The audio data may include an audio signal component and a background noise component. As a subset of the audio data is received, the processor may identify based on at least the subset of audio data, the one or more error conditions. The processor may then cause a graphical display to display a graphical representation associated with the identified error condition.

Abstract: An impulse response between a sound-producing source and a listening point in an acoustic space is measured. A frequency characteristic serving as the processing target of sound field correction is derived from the impulse response. A level difference at a boundary frequency between a level representing a low frequency band and a level representing middle and high frequency bands is calculated for the low frequency band equal to or lower than the boundary frequency and the middle and high frequency bands higher than the boundary frequency in the frequency characteristic. The level of a target characteristic in the low frequency band in the frequency characteristic is decided to set the level difference after sound field correction to be equal to or smaller than a predetermined value.

Abstract: A loudspeaker that measures the impulse response of a listening area is described. The loudspeaker may output sounds corresponding to a segment of an audio signal. The sounds are sensed by a listening device proximate to a listener and transmitted to the loudspeaker. The loudspeaker includes an adaptive filter that estimates the impulse response of the listening area based on the signal segment. An error unit analyzes the estimated impulse response together with the sensed audio signal received from the listening device to determine the accuracy of the estimate. New estimates may be generated by the adaptive filter until an accuracy level is achieved for the signal segment. A processor may utilize one or more estimated impulse responses corresponding to various signal segments that cover a defined frequency spectrum for adjusting the audio signal to compensate for the impulse response of the listening area. Other embodiments are also described.

Abstract: Processor-implemented methods and systems for visually-assisted mixing of audio using a spectral analyzer are disclosed. The system calculates and displays a spectral view for each track in an arrangement in a multi-track view. A user can then request modification of the spectral view. In response to this request for modification, the system automatically adjusts associated mixing parameters for the modified track so that the spectral output of the track substantially matches the user-requested modified spectral view. This allows a user to visually mix an arrangement by imputing a desired spectral result and having the program make the changes necessary to achieve it.

Abstract: A spectral response shifting method allows a listener to shift a system response to match their preferences. The method includes a lookup table of coefficients for a number of Infinite Impulse Response (IIR) filter or Finite Impulse Response (FIR) filter which are selected by the listener to shift the response either toward bass or towards treble. In one embodiment, 5th order IIR biquads filters are used at ½ dB increments from ?6 dB to +6 dB. For example, when a listener selects the “more treble by 1 dB” setting, a set of coefficients is loaded providing ?1 dB in bass cut at the lowest frequency of the audible spectrum (20 Hz) and +1 dB of treble boost at the highest frequency of the audible spectrum (20 kHz). The frequency response at all other points is defined by interpolating between the highest frequency (20 kHz) and the lowest frequency (20 Hz).

Abstract: An approach is provided for providing audio-based control for at least one device based on changes in the rendering of audio information. A control platform causes, at least in part, a transmission of audio information from at least one device to an audio playback assembly for rendering. The control platform further determines at least one indication of a change in the rendering, wherein the at least one indication is based, at least in part, on one or more controls associated with the audio playback assembly. The control platform also processes and/or facilitates a processing of the at least one indication to cause, at least in part, an execution of at least one function for controlling the rendering of the audio information.

Abstract: In Higher Order Ambisonics, a problem is the tracking of time variant directions of dominant sound sources. The following processing is carried out: from a current time frame of HOA coefficients, estimating a directional power distribution of dominant sound sources, from said directional power distribution and from an a-priori probability function for dominant sound source directions, computing an a-posteriori probability function for said dominant sound source directions, depending on said a-posteriori probability function and on dominant sound source directions for the previous time frame, searching and assigning dominant sound source directions for said current time frame of said HOA coefficients.

Abstract: Systems and method are disclosed for facilitating efficient calibration of filters for correcting room and/or speaker-based distortion and/or binaural imbalances in audio reproduction, and/or for producing three-dimensional sound in stereo system environments. According to some embodiments, using a portable device such as a smartphone or tablet, a user can calibrate speakers by initiating playback of a test signal, detecting playback of the test signal with the portable device's microphone, and repeating this process for a number of speakers and/or device positions (e.g., next to each of the user's ears). A comparison can be made between the test signal and the detected signal, and this can be used to more precisely calibrate rendering of future signals by the speakers.

Abstract: Systems and methods for removing or reducing distortion in an ultrasonic audio system can include receiving a first audio signal, wherein the first audio signal represents audio content to be reproduced using the ultrasonic audio system; calculating a first error function for the ultrasonic audio system, the first error function comprising an estimate of distortion introduced by reproduction of the audio content by the ultrasonic audio system; transforming the first audio signal into a first pre-conditioned audio signal by combining the first error function with the first audio signal; and modulating the transformed audio signal onto an ultrasonic carrier.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal.

Abstract: An apparatus and method for keeping output loudness and quality of sound in different equalizer modes. The apparatus includes a gain setting unit configured to set a protection limit of an output voltage of an input audio signal; an equalizer (EQ) mode filter configured to filter the input audio signal so as to achieve the different equalizer modes; an amplitude real-time processing module configured to receive the audio signal output from the EQ mode filter or an audio signal feedback, and process the received audio signal such that the output voltage of the processed audio signal is lower than the protection limit; and a digital-to-analog converter configured to perform a digital-to-analog conversion on the processed audio signal, and output an analog audio signal.

Abstract: A networked-speaker system which includes noise cancelling for active suppression of unwanted sounds. Feed-forward or feed-back cancelation can be used as appropriate for the location of the microphone sensing the noise.

Abstract: A three channel speaker system is provided that includes a three-channel processor to receive a signal from an audio source, where the speaker system includes a center-channel speaker on a first face, a left-channel speaker on a second face, and a right-channel speaker on a third face, where the first face further includes first and second passive radiators positioned on opposite sides of the center-channel speaker, and also includes a fourth face comprising third and fourth passive radiators. Where all three-channel speakers are on a front face, passive radiators may be positioned in between each of the speakers.

Abstract: A system includes a plurality of channels, a parameter calculator, and an equalizer. Each channel receives a plurality of input signals that represent an input to a speaker. Each channel generates real and imaginary components of a transfer function of the speaker based on the plurality of input signals. The parameter calculator calculates a plurality of parameters of the speaker based on the real and imaginary components of the transfer function received from each channel. The equalizer adjusts the input to the speaker based on the parameters.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal and setting a peak signal threshold to a level that is determined to be safe, particularly for children or listeners with sensitive hearing. In various embodiments, a method comprises adjusting a gain of the signal based upon a received profile value minus a safety parameter, filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that sets a peak threshold based upon the profile value minus the safety parameter, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer, passing the signal through a second compressor that sets a peak threshold based upon the profile minus the safety parameter, and outputting the signal.

Abstract: According to one embodiment, an acoustic signal corrector includes: an output module; a selection receiver; and a holder. The output module is configured to output a plurality of acoustic signals. Amplitude values of frequencies within a frequency band of each of the acoustic signals are emphasized as emphasized amplitude values. A plurality of amplitude values among the emphasized amplitude values are corrected as corrected amplitude values at some of frequencies within the frequency band. Resonance is possibly induced in the frequency band by sealing an ear canal. The selection receiver is configured to receive a selection of one of the acoustic signals output by the output module. The holder is configured to hold, as a configuration for sound quality correction, a configuration corresponding to the correction of the one of the acoustic signals at the some of the frequencies.

Abstract: An audio signal processing apparatus includes a signal processing unit, an output unit, a retention unit, and a coefficient setting unit. The signal processing unit is configured to perform signal processing on an audio signal by a digital filter. The output unit is configured to be connected to an external speaker and output the audio signal to the speaker. The retention unit is configured to retain a plurality of filter coefficients that are impulse responses having reverse characteristics of a plurality of speakers having different speaker characteristics. The coefficient setting unit is configured to select one of the filter coefficients that corresponds to the speaker connected to the output unit from the retention unit and set the filter coefficient in the digital filter.

Abstract: A software-based equalization method corrects for effects of impedance interactions of a source device and an output device, on an aggregate frequency response, in audio playback of a system. The correction results in perceived sound remaining consistent from any source device and output device combination. The method does not require any hardware modifications and a cloud-based database of source devices and output devices provides corrections for a wide array of possible device pairings.

Abstract: A hearing aid device with a dichotic function comprises a first sound input/output component having a first microphone configured to be worn on one ear, a second sound input/output component having a second microphone configured to be worn on the other ear, and a dichotic setting component. The dichotic setting component determines the occurrence of time masking according to a sound signal inputted from the first microphone or the second microphone, and activates or deactivates the dichotic function according to this determination result. The hearing aid device improves the hearing aid effect.

Abstract: A control device and an associated method for driving a multi-function speaker supporting a plurality of predetermined functions including at least an audio function and a non-audio function includes a digital signal mixing block and a digital-to-analog block. The digital signal mixing block is arranged for receiving a plurality of digital input signals corresponding to the predetermined functions, respectively, and generating a digital mixed signal according to the digital input signals. The digital-to-analog block is coupled to the digital signal mixing block, for generating an analog driving signal to the multi-function speaker according to the digital mixed signal.

Abstract: Described are systems and methods performed by computer to reduce crosstalk produced by loudspeakers when rendering binaural sound that is emitted from the loudspeakers into a room. The room may have sound-reflecting surfaces that reflect some of the sound produced by the loudspeakers. To reduce crosstalk, a room model stored by the computer, is accessed. The room model models at least sound reflected by one or more of the physical surfaces. The room model is used to calculate a model of an audio channel from the loudspeakers to a listener. The model of the audio channel models sound transmission from the loudspeakers to the listener. The computer uses the model of the audio channel to cancel crosstalk from the loudspeakers when rendering the binaural sound.

Abstract: An audio signal processing device includes two audio signal processing units that serially perform a processing with respect to an input audio signal, and obtain an output audio signal for driving a speaker. One audio signal processing unit of the two audio signal processing units performs, with respect to the input audio signal, a correction process through a filter that realizes a reverse characteristic of an impulse response measured at a first measurement position that is a front position of the speaker. The other audio signal processing unit performs, with respect to the input audio signal, a correction process through a filter that realizes a reverse characteristic of an impulse response measured at a second measurement position different from the first measurement position that is a front position of the speaker.

Abstract: The invention relates to a process for determining an averaged frequency-dependent transfer function for a perturbed linear time-invariant system by means of an evaluation device, wherein the process comprises providing frequency-dependent reference signals derived from excitations input in a linear time-invariant system, providing frequency-dependent measuring signals for the linear time-invariant system associated with the frequency-dependent reference signals, and determining an averaged frequency-dependent transfer function for the linear time-invariant system, in that, using signal deconvolution of mutually associated measuring and reference signals, frequency-dependent transfer functions are determined and the frequency-dependent transfer functions are averaged, and wherein during determination of the averaged frequency-dependent transfer function at least a part of the determined frequency-dependent transfer functions is included in the averaging corresponding to a respectively associated frequency-dep

Abstract: A signal processing system for generating an ultrasonic signal from an audio signal comprises a compressor, operable to compress the audio signal, and an equalization circuit, operable to equalize the audio signal. A modulation circuit is operable to combine the audio signal with a carrier signal to produce at least one modulated carrier signal. A voltage detection and control circuit is operable to: detect a voltage of the modulated carrier signal; compare the detected voltage of the modulated carrier signal to a desired voltage; and cause the voltage of the modulated carrier signal to be adjusted if the compared voltage differs by a predetermined amount from the desired voltage.

Abstract: The invention relates to a method and a loudspeaker for storing equalization settings in an active loudspeaker including a loudspeaker cabinet, at least one driver connected to the cabinet and an amplifier electronics including an equalizer, in which method the loudspeaker is calibrated in controlled conditions for forming factory calibration parameters for controlling the equalizer, the factory calibration parameters controlling the equalizer are stored in the amplifier electronics. In accordance with the invention the factory calibration parameters are also stored in a memory fixedly mounted to the loudspeaker.

Abstract: A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation.

Abstract: A system for headphone equalization includes a stored set of predetermined tone burst reference signals and a stored set of predetermined tone burst test signals that form a range of frequencies used in a user specific audio test to develop a headphone correction filter. A predetermined tone burst reference signal and a predetermined tone burst test signal may intermittently and sequentially drive a transducer included in the headphone. A loudness of the predetermined tone burst reference signal may be fixed and a loudness of the predetermined tone burst test signal may be variable with a gain setting. The gain setting may be used to generate the headphone correction filter.

Abstract: Apparatus and methods are disclosed for acoustic optimization. An example playback device includes a first transducer to at least one of output sound waves and receive sound waves, a second transducer to at least one of output sound waves and receive sound waves, and an acoustic grille positioned in relation to the first transducer, where the acoustic grille is to reflect sound waves received at a first angle of incidence.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: Frequency equalization system substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range. The frequency equalization system uses multiple microphones to measure the impulse responses of the room and uses the impulse responses to design filters to process the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features very short computation time.

Abstract: According to one embodiment, audio controlling apparatus includes first receiver configured to receive audio signal, second receiver configured to receive environmental sound, masking gain calculator configured to calculate masking gain for each frequency band, based on audio signal and environmental sound, and gain controller configured to smooth masking gain of frequency band that is less than preset threshold frequency in first interval, smooth masking gain of frequency band that is equal to or higher than threshold frequency in second interval that is longer than first interval, and thereby set gain.

Abstract: A signal processing system for generating a parametric signal comprises an audio compressor, operable to compress a dynamic range of an audio input signal, and an equalization network, operable to equalize the audio signal. A low pass filter is operable to remove high portions of the audio signal and a high pass filter is operable to remove low portions of the audio signal. An oscillator circuit is operable to generate a carrier signal, and a modulation circuit is operable to combine the audio signal with the carrier signal to produce at least one modulated carrier signal.

Abstract: Signal processing section of a terminal converts acquired audio signals of a plurality of channels into frequency spectra set, calculates sound image positions corresponding to individual frequency components, and displays, on a display screen, the calculated sound image positions results by use of a coordinate system having coordinate axes of the frequency components and sound image positions. User-designated partial region of the coordinate system is set as a designated region and an amplitude-level adjusting amount is set for the designated region, so that the signal processing section adjusts amplitude levels of frequency components included in the frequency spectra and in the designated region, converts the adjusted frequency components into audio signals and outputs the converted audio signals.

Abstract: Methods and systems are provided for, volume interactions for connected playback device. In one example, a plurality of playback devices may be rendering audio in synchrony, when a volume adjustment is made for the plurality of playback devices as a group. In one case, a first playback device in the plurality of playback devices may have a volume limit, as indicated by a state variable associated with the first playback device. In this case, the resulting volume adjustment for the first playback device may be made according to both the group volume adjustment and the volume limit of the first playback device.

Abstract: Disclosed are methods and devices for adapting a ringtone in accordance with ambient noise. In one embodiment a method includes processing an ambient noise signal to determine an ambient noise volume. When there is an incoming communication and it is determined that the ambient noise has a low ambient noise volume, a method can include generating a ringtone having a ringtone volume that is initially a low volume and increasing the ringtone volume over a predetermined period of time. A method can include applying a filter to the ringtone signal to increase a signal-to-noise ratio, the filter configured to increase an amplitude of a frequency that is not one of the predominant frequencies of the ambient noise and that the transducer is configured to output according to the transducer frequency response. In another embodiment, a second transducer may generate a ringtone output in addition to the first transducer.

Abstract: Systems and methods to determine a position of a base with respect to a playback device and evaluating whether the position is proper or improper given an orientation of the playback device are disclosed and described herein. In one aspect, a playback device is provided. The example playback device includes a first fastener located on a first side of the playback device, a processor, and memory having stored thereon instructions executable by the processor to cause the computing device to perform functions. The functions include determining that a base is not coupled to the playback device via the first fastener and, in response to determining that the base is not coupled to the playback device via the first fastener, adjusting a playback setting of the playback device. The functions further include facilitating output of content via the playback device according to the adjusted playback setting.

Abstract: Embodiments described herein provide for acoustic signatures in a playback system. In some embodiments, a playback device includes an audio processing component configured to receive an audio input signal, receive an acoustic signature signal, generate an audio output signal based on the audio input signal and the acoustic signature signal, and output the generated audio output signal. In some embodiments, the acoustic signature identifies at least one of the playback device or a group of playback devices. Some embodiments additionally include the playback device amplifying the generated audio output signal with one or more amplifiers, and playing the generated audio output signal via one or more speakers.

Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the M RI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: A microphone includes a first signal input, a second signal input and a control unit coupled to the first signal input and the second signal input. The control unit is configured to, upon receiving a support signal at one of the first signal input and second signal input, process a signal received at the other of the first signal input and second signal input as an incoming data signal.

Abstract: A method of attenuating an input signal to obtain an output signal is described. The method comprises receiving the input signal, attenuating the input signal with a gain factor to obtain the output signal, applying a filter having a frequency response with a frequency-dependent filter gain to at least one of a copy of the input signal and a copy of the output signal to obtain a filtered signal, the frequency-dependent filter gain being arranged to emphasize frequencies within a number N of predetermined frequency ranges, N>1; wherein the filter comprises a sequence of N sub-filters, each one of the N sub-filters having a frequency response adapted to emphasize frequencies within a corresponding one of the N predetermined frequency ranges; determining a signal strength of the filtered signal, and determining the gain factor from at least the signal strength.

Abstract: An echo caused by sound leakage from a housing that vibrates due to a vibrating body is reduced. An electronic device (1) according to the present invention includes a piezoelectric element (30), a vibrating plate (10) that vibrates due to the piezoelectric element (30), microphones (42, 43), and an equalizer (44), the electronic device (1) causing the vibrating plate (10) to generate air-conducted sound and vibration sound that is transmitted by vibrating a part of a human body. The equalizer (44) makes a low-range emphasis setting, to emphasize a low frequency range more than a high frequency range of the air-conducted sound, when the volume of the air-conducted sound collected by the microphones (42, 43) exceeds a predetermined threshold.

Abstract: The application discusses a computer implemented method and apparatus for performing audio equalisation in an audio receiver device, such as an integrated receiver/decoder or set top box, or integrated TV, connected to one or more audio playback devices, such as a television unit, computer screen and speakers, amplifier or home theatre equipment. The method and apparatus use an equalisation process which compares audio signals received in different audio formats (e.g. MPEG-1 Layer II, AC-3 2.0, AC-3 5.1 and HE-AAC) with one another, allowing a correction gain factor to be determined for equalising the perceived loudness of the signals when played-back at a connected playback device. The correction gain factor is then applied in the audio receiver device before output.

Abstract: A filter for equalizing the frequency response of loudspeaker systems includes at least one band filter section (11) comprised an n-order high boost or cut shelving fitter (13) having a break point frequency, ?1, and an n-order low boost or cut shelving filter (15) having a break point frequency, ?2, wherein ?1<?2. The order, n, of at least one, and preferably both of the shelving filters of the band filter sections can he selected for adjusting the slope of the shelving filter at one or both of its break point frequencies. The high and low n-order shelving filters forming the band filter sections have substantially the same gain and produce a resultant band gain for the band filter section. Gain correction is provided for the selectable n-order high shelving filter and n-order low shelving filter for correcting the resultant band gain to a base gain level.

Abstract: According to one embodiment, a characteristic correcting device includes: a correction filter configured to correct sound quality characteristics of a plurality of bands in a frequency range of an input signal based on a frequency characteristic which is set in advance to generate an output signal; an input module configured to input a surrounding sound signal of sound around an output device outputting the output signal; and an adjusting module configured to reduce the number of the bands of which the sound quality characteristics are to be corrected of the bands in the frequency range of the input signal in accordance with an increase in amplitude of the surrounding sound signal which is input.

Abstract: A headphone comprises a plurality of actuatable equalization selectors. Each of the selectors corresponds to an equalization setting that includes a preset distribution of relative amplitudes of sounds in predetermined frequency ranges. In one embodiment, each of the plurality of actuatable equalization selectors is a button-type switch. A knob-type switch or a voice recognition mechanism could also actuate an equalization setting. In a preferred embodiment, an equalizer identification indicator produces a communication perceivable to a headphone wearer and which corresponds to an equalization setting. The communication can be audible, preferably a human voice, or tactile, preferably vibration patterns corresponding to equalization settings.

Abstract: A channel divider that can suitably set a crossover frequency when a biwiring-connectable multiway speaker system including network circuits are used is provided. The channel divider includes a low-pass filter LPF and a high-pass filter HPF for dividing a band of a sound signal into a first output signal and a second output signal so as to output the signals, and a control circuit for controlling the filters. The channel divider has a sound output mode in which the control circuit sets a cutoff frequency fcL of LPF to a value higher than fc0 by about ? to 1 Oct., sets a cutoff frequency fcH of HPF to a value lower than fc0 by about ? to 1 Oct., and outputs the first output signal and the second output signal including frequency bands fcL to fcH when any crossover frequency fc0 for defining the band division is specified.

Abstract: An apparatus including at least one sensor configured to sense location of at least one portion of a head of a user relative to the apparatus; at least one processor; and at least one memory having software. The processor and the software are configured to process audio signals based, at least partially, upon output from the sensor.

Abstract: In a multiband compressor 100, a level calculation unit 121 calculates a signal level inputted for each of bands, a gain calculation unit 122 calculates a gain value from the calculated signal level, and a gain limitation unit 130 limits a gain value by comparison with a gain value of the other band in a compressor for each band. With this configuration, provided is a multiband compressor capable of achieving a balance between the quality of sound and the effect of enhancing the sound level at a high level.