Abstract: A method is provided for controlling a dynamic range of audio reproduction in an interior of a vehicle. An input audio signal is received by an audio processor. The input audio signal is split into a first and a second processing path. The input audio signal of the second processing path is compressed. A noise-related parameter of the vehicle is characterized. A gain control is applied to the compressed audio signal of the second processing path in response to the noise-related parameter. The compressed and uncompressed audio signals are synchronously recombined. The combined audio signal is reproduced to the interior of the vehicle.

Abstract: The present invention(s) provides for methods and systems for digitally processing a ringtone. In various embodiments, a method comprises generating a profile comprising a plurality of filter equalizing coefficients, configuring a plurality of filters of a graphic equalizer using the plurality of filter equalizing coefficients from the profile, receiving a ringtone for processing, adjusting the plurality of filters using a first gain, equalizing the ringtone using the plurality of filters of the graphic equalizer, and outputting the equalized ringtone.

Abstract: Techniques and systems for enhancing delivered audio signals are disclosed which may be employed in a delivery system at a server side, a client side, or both. The techniques include forming a processed audio signal by processing audio signals through multiple pathways which operate on different frequency bands using dynamic processing and other elements, and thereafter providing recording or listening environment enhancements and other sound enhancements to the processed audio signal. Also disclosed are techniques and systems for implementing the multi-pathway processing and environmental and sound enhancements.

Abstract: A computer-implemented method and system are provided for profiling or analyzing audio data based on an audio parameter. The audio data is divided into audio data into segments. Each segment is characterized by the audio parameter in a defined range. Effects are selected according to the segments meeting criteria for the audio parameter of the effects. A media presentation is authored or generated using the selected effects.

Abstract: An apparatus for processing an audio input signal is provided and includes an audio processing circuit and an audio compressing circuit. The audio processing circuit receives the audio input signal, and enhances a first frequency part of the audio input signal to output a bass-enhancement signal. The audio compressing circuit is coupled to the audio processing circuit, and reduces a gain of a second frequency part of the bass-enhancement signal to output an audio output signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and preset information; obtaining preset matrix from the preset information, wherein the preset matrix indicates contribution degree of the object to output channel; and adjusting output level of the object by using the preset matrix. Accordingly, without user's setting for each object, if preset metadata to be applied to an audio signal is selected with reference to previously-set preset metadata, levels of objects included in the audio signal can be easily adjusted using preset rendering data corresponding to the selected preset metadata.

Abstract: The invention concerns a method and a system for sound spatialization of a first set of not less than one of the audio channels encoded on of a number of frequency subbands (SBk) and decoded in a transformed domain (Fl, C, Fr, Sr, SI, Ife) into a second set of not less than two (Bl, Br) sound channels in the time domain, from modelling filters converted into a gain and a delay applicable in the transformed domain involving: filtering (A) through equalization, subband delay of the signal by applying at least one gain and one delay to generate from each of said encoded channels an equalized and delayed component; adding (B) a subset of equalized and delayed signals to create a number of filtered signals corresponding to not less than two; synthesizing (C) each of said filtered signals to obtain the second set of not less than two reproduction sound channels (Bl, Br) in the time domain.

Abstract: Techniques pertaining to providing an appropriate recording mode to achieve optimum record data in accordance with the recording environment are described. In one embodiment, a current audio recording mode is initially selected to be one of a low sensitivity audio recording mode and a high sensitivity audio recording mode. A digital gain for digital audio recording data in the current audio recording mode is obtained. The current audio recording mode is automatically switched between the low sensitivity audio recording mode and the high sensitivity audio recording mode when the digital gain reaches a gain threshold.

Abstract: A multiband DRC system and a method for controlling the same are disclosed. In accordance with the present invention, the system and the method are capable of increasing an overall loudness of an output signal by controlling thresholds and gains of plurality of DRCs included in the multiband DRC system according to frequency bands.

Abstract: Systems and method for adapting a gain of an audio signal containing at least two different tracks with different signal level ranges. An example system includes a loudness determination unit for determining a perceived loudness of the audio input signal based on a psychoacoustic model of a human hearing. The system further includes a loudness adaptation unit configured to determine a loudness adaptation indicator based on the perceived loudness and a predetermined loudness, and to introduce the loudness adaptation indicator into the audio signal to generate an adapted gain signal. A gain determination unit is configured to adapt the gain for outputting the audio signal containing the at least two tracks based on the loudness adaptation indicators in the adapted gain signal at about the predetermined loudness.

Abstract: A multimedia controller including a processor, the controller configured to: provide a user interface for a player group, wherein the player group includes a plurality of players in a local area network, and wherein each player is configured to playback a multimedia output from a multimedia source; accept an input to facilitate formation of the player group, indicating that at least two of the players in the local area network are to be included in the player group; for each of the plurality of players within the player group, accept an input to adjust a volume associated with the player, that causes the player to adjust its volume; and accept an input to adjust a volume associated with the player group, wherein the input to adjust the volume associated with the group causes the players in the player group to adjust their volumes.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

Abstract: A loudspeaker drive circuit comprises a signal path compressor/limiter (12) for implementing a change to the peak-mean amplitude ration in the time domain. A feedforward control loop measures an acoustic quality of the signal at the output of a control loop compressor/limiter (20) or the output of the signal path compressor/limiter (12), and also estimates a loudspeaker excursion based on the signal at the output of the control loop compressor/limiter (20). The signal path compressor/limiter is controlled based on the acoustic quality measurement and excursion estimation. The invention provides a method for the maximisation of the acoustic output of a loudspeaker by adjusting the characteristics of a compressor/limiter, with the constraint that the audio quality stays acceptable, and that the diaphragm displacement does not exceed a certain threshold.

Abstract: A volume control system comprises: an amplifying apparatus; and a controller connectable to the amplifying apparatus. The amplifying apparatus includes; a first receiving section for receiving a volume value from the controller, a first input section into which an instruction for changing the volume value is input by a user's operation, a first transmitting section for transmitting the volume value input into the first input section to the controller, and a volume control section for, when the first receiving section receives the volume value or the changed volume value is input into the first input section, controlling a volume value of the amplifying apparatus based on the volume value.

Abstract: A technique is disclosed to implement crossfading of audio tracks. In one embodiment, the function describing the fade out of the ending audio track and/or the slope describing the fade in of the beginning audio track may be altered to increase the perceptible overlap of the two tracks. In another embodiment, the duration of the fade out and/or of the fade in may be altered to increase the perceptible overlap of the two tracks. In other embodiments, one or both of the function and/or duration of the fade out and/or fade in effect may be altered to improve the perceptibility of the overlap or the audio tracks.

Abstract: An apparatus is provided for remotely monitoring a plurality of amplifiers on a display device. The amplifiers are provided to process audio signals and directly or indirectly connected to a network. In the apparatus, a collecting section collects a group code from each of the amplifiers through the network. The group code is allocated to each amplifier according to a grouping system for grouping the plurality of the amplifiers. A display control section displays a tree diagram of the amplifiers according to the collected group codes on the display device as a graphical representation of the grouping system.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract: A volume control apparatus that is connectable to a controller and controls a volume value of a sound signal based on an instruction from the controller includes a receiving section for, when a controller-side volume setting value settable in the controller is changed by a user's operation, receiving the changed controller-side volume setting value from the controller. The apparatus also includes a first difference value calculating section for calculating a first difference value between the changed controller-side volume setting value and the controller-side volume setting value before change, a second difference value calculating section for calculating a second difference value between a maximum volume value controllable by the controller and a current volume value, and an increase value calculating section for multiplying the second difference value by a ratio of the first difference value so as to calculate an increase value of the volume value.

Abstract: Information useful for modifying the dynamics of an audio signal is derived from one or more devices or processes operating at one or more respective nodes of each of a plurality of hierarchy levels, each hierarchical level having one or more nodes, in which the one or more devices or processes operating at each hierarchical level takes a measure of one or more characteristics of the audio signal such that the one or more devices or processes operating at each successively lower hierarchical level takes a measure of one or more characteristics of progressively smaller subdivisions of the audio signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: A system includes a plurality of inputs each configured to receive a filtered version of a source signal. The system extracts the energy information from each input signal and compares the energy information of a plurality of input signals. Alternatively, the system extracts energy information from a signal that is the difference of two input signals. Based on the energy information, the system determines at least one parameter that may be changed in at least one circuit in a plurality of circuits to minimize the differences in energy of the input signals or to minimize the energy of the difference signal. Parameters may include for example amplification, delay, and corner frequency values. The set of circuits may include microphone interface circuits. Merely by way of example, a system with microphone interface circuits may be included in a hearing enhancement device or in a hands-free earpiece.

Abstract: An electronic apparatus such as an audio apparatus, television signal receiver, or other apparatus provides an equalizer interface that enables equalizer settings to be adjusted via acoustic user feedback. According to an exemplary embodiment, the electronic apparatus includes a sound generator for generating a first tone for a first frequency band. A processor enables a user to adjust a first volume level of the first tone. A memory stores the first volume level for the first frequency band. A volume controller applies the first volume level to output signals in the first frequency band.

Abstract: Systems and method for adapting a gain of an audio signal containing at least two different tracks with different signal level ranges. An example system includes a loudness determination unit for determining a perceived loudness of the audio input signal based on a psychoacoustic model of a human hearing. The system further includes a loudness adaptation unit configured to determine a loudness adaptation indicator based on the perceived loudness and a predetermined loudness, and to introduce the loudness adaptation indicator into the audio signal to generate an adapted gain signal. A gain determination unit is configured to adapt the gain for outputting the audio signal containing the at least two tracks based on the loudness adaptation indicators in the adapted gain signal at about the predetermined loudness.

Abstract: A method and apparatus of an adaptive gain control (AGC) unit. The method includes receiving a noisy input signal and determining to utilize a stethoscope in at least one of a noise suppression mode or in amplification mode depending if the noise level is at least one of above or below a threshold, wherein the stethoscope is in noise suppression mode when K is less than the threshold and is amplification mode when K is above the threshold.

Abstract: The invention relates to a compressor and method for amplifying an input signal with a controlled gain. An output signal representing the input signal is amplified by an initial gain and a signal level of the input signal or of the output signal is compared with a threshold level. If the signal level is below the threshold level, the initial gain value is updated using an adaptive control characteristic, and if the signal level is above the threshold level, the initial gain value is updated using a fixed control characteristic or an adaptive control characteristic respectively. The adaptive control characteristic is dependent on the signal level and the fixed control characteristic is independent from the signal level.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: An audio processing chip includes a connecting port, an audio amplifier module and a pulse width modulation (PWM) control circuit. The connecting port receives a pulse width modulation (PWM) signal; the audio amplifier module amplifies an audio signal according to a control signal to thereby output an audio output signal; and the pulse width modulation (PWM) control circuit is coupled between the connecting port and the audio amplifier module, and outputs the control signal to the audio amplifier module according to the PWM signal to thereby control an operation of the audio amplifier module.

Abstract: An input frame data producing unit produces from data stored in an input buffer input frames each including a predetermined number of sub-frames of a first hopsize determined based on the first frame size and the overlapping rate. A frame processing unit executes a window function on the input frames and shifts the windowed input frames by the first hopsize and overlaps the shifted input frames, storing the overlapped frames in an output frame. An output buffer data producing frame unit stores data from the output frame to an output buffer including a predetermined number of sub-frames of a second hopsize. A CPU sets the first hopsize and overlapping rate in a slow-speed reproduction when the reproducing speed ratio is set lower than 1 different from in a high-speed reproduction when the reproducing speed ratio is set larger than 1.

Abstract: In accordance with the invention, audio signals are specially processed for sound presentation in a high noise environment. The electrical signal representative of the sound is first subjected to equalization to preferentially reduce the magnitude of bass signals while increasing the magnitude of treble signals. The equalized signal is then compressed, and the compressed signal is subjected to “mirror image” equalization which increases the magnitude of bass signals while reducing the magnitude of treble signals. The resulting signal fed to the speakers provides a sound presentation of compressed volume range and a bass-rich sound spectrum. It is particularly useful for providing quality sound presentation in a high noise environment.

Abstract: An audio compressor may regulate the level of an audio input signal depending on whether the level is above or below a threshold value. The audio compressor may control a pumping that is created when regulating a dynamic range of an audio signal with the threshold value. A feedback loop connecting the signal output from the audio compressor with a release filter may be used to modify an effective release time of the signal. A controller may be used that allows a filter coefficient of the release filter to be controlled to adjust the effective release time as a function of the signal output.

Abstract: A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The apparatus comprises an information receiving unit receiving a downmix signal including and a plurality of preset informations; an external preset information receiving unit receiving a plurality of external preset informations being inputted from external; an external preset applying-determining unit determining whether the plurality of the external preset informations applies to the downmix signal; an external preset information selecting unit selecting one external preset information, if the plurality of external preset informations is selected; and a rendering unit controlling the object by applying the external preset information to the all data regions. Accordingly, an audio signal can be efficiently reconstructed by individually selecting and applying external preset information by a data region unit or by selecting and applying the same external preset information to a whole downmix signal.

Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

Abstract: A modified limiter of the present invention produces a consistent loudness level across one or more input tracks. The modified limiter has a target threshold, wherein signal levels below the target threshold are adjusted. The signal levels below the target threshold are adjusted by using an upward expander, which is not a time-based parameter but, instead, is based on signal levels only. A release target of the modified limiter is a highest peak within a look-ahead delay. The modified limiter is highly optimized as compared to other look-ahead limiters. As such, its processing footprint on an application such as audio management software like iTunes™, or a consumer music playback device like an iPod™, is minimal. The modified limiter is also able to be used in other technologies, such as but not limited to cell phones like an iPhone™ and set-top boxes like an Apple TV™.

Abstract: A method for controlling consumer exposure to audio dosage for a composite audio signal, the method comprising: sampling the composite audio signal; transforming the sampled audio signal, using a fast fourier transform algorithm, to produce a signal representative of the amplitude of component frequencies of the audio signal; comparing the transformed audio signal with a predefined impulse control threshold profile, representing the target maximum amplitude for each frequency component for the audio signal threshold profile to produce a configuring signal representative of the difference between the broadcast signal and the profile; using the configuring signal to automatically configure in real time a Finite Impulse Response (FIR) filter so that it attenuates the amplitude of the transformed audio signal in frequency bands centered on the frequencies at which the target threshold is exceeded; and outputting the attenuated audio signal for consumer exposure.

Abstract: A signal processing apparatus includes: clip detector means that detects presence/absence of a clipped part with a deformed waveform in each of N audio signals output from N microphones (where N is an integer equal to or greater than 2) based on a dynamic range of a circuit; and interpolation means that treats an audio signal in the N audio signals which has the clipped part detected by the clip detector means as an interpolation target, and other audio signals as non-interpolation targets, and interpolates the waveform of the clipped part of the interpolation target using the waveform of at least one audio signal in the non-interpolation targets.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: An apparatus and a method for automatically controlling a gain using phase information are provided. The apparatus includes a frequency conversion unit converting each of input signals received from a plurality of acoustic input apparatuses to frequency input signals, a factor determination unit determining a scaling factor according to a gain difference between the input signals based on a phase difference of the frequency input signals, and a scaling performance unit performing scaling for gain compensation between the input signals based on the scaling factor.

Abstract: A method to audio matrix encode/decode, which encode and decode audio signals of two or more channels into an audio signal of one or more channel while preserving the direction of a sound image includes extracting pieces of sound image information from audio signals of multi channels, encoding and allocating the extracted sound image information to an inaudible frequency domain except an audible frequency domain, and adding the sound image information allocated to the inaudible frequency domain and matrix-encoded stereo signals of the audible frequency domain.

Abstract: A system for generating parametric stereo data from phase modulated stereo data is provided. A phase difference system receives left channel data and right channel data and determines a phase difference between the left channel data and the right channel data. A phase difference weighting system receives the phase difference data and generates weighting data to adjust left channel amplitude data and right channel amplitude data based on the phase difference data. A magnitude modification system adjusts the left channel amplitude data and the right channel amplitude data using the weighting data to eliminate phase data in the left channel data and the right channel data.

Abstract: An acoustic shock protection method and device are provided. A pattern analysis-based approach is taken to an input signal to perform feature extraction. A parameter space is identified, which is corresponding to, the signal space of the input signal. A rule-based decision approach is taken to the parameter space to detect an acoustic shock event. The device may be advantageously implemented using a weighted overlap-add approach to provide low group delay, high-fidelity and a high degree of protection from acoustic shock events.

Abstract: A device and method are provided for automatically adjusting gain, including a conversion module for converting an audio time-domain signal to an audio frequency-domain signal, an analysis module for analyzing the audio frequency-domain signal in accordance with an equal-loudness level contour of human hearing so as to generate strength weightings and generating a signal strength in accordance with the weightings, a calculation module for calculating a gain by analysis of the audio frequency-domain signal when the signal strength falls outside a default range, and a control module for generating an audio output signal in accordance with the gain and the audio time-domain signal.

Abstract: A method (400, 500) and apparatus (220) seeks to improve the intelligibility of speech emitted into a noisy environment. Formants are identified (426) and perceptual frequency scale band is selected (502) that includes at least one of the identified formants. The SNR in each band is compared (504) to a threshold and, if the SNR for that band is less than the threshold, the method increases a formant enhancement gain for that band. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains yielding combined gains that are then clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532), and used to reconstruct (532, 534) an audio signal.

Abstract: An audio signal processing apparatus, includes an environmental ambient noise level detection unit for detecting an environmental ambient noise level contained in an audio signal inputted through sound collection means for collecting a transmission sound at the time of a voice call, a signal level adjustment unit which has a level adjustment function to adjust an output signal level with respect to an input signal level, and an input/output characteristic change function to change an input/output characteristic when adjusting a level in the level adjustment function by means of a control signal, and in which a received sound signal in the case of the telephone call voice is arranged to be an input signal, and a control signal generation unit for generating the control signal for changing the input/output characteristic of the signal level adjustment unit from the environmental ambient noise level detected by the environmental ambient noise level detection unit.

Abstract: A system for limiting the excursion of an audio speaker. The system may include first lowpass filter circuitry configured to receive a system input signal and to generate a first signal. The system may further include clipping circuitry configured between the first lowpass filter circuitry and the first combiner circuitry. The system may include first highpass filter circuitry configured to receive the system input signal and to generate a second signal. The system may include second lowpass filter circuitry configured to receive a combined output signal from the first combiner circuitry. The system may include allpass filter circuitry configured to receive the system input signal and to generate a fourth signal. The system may include second highpass filter circuitry configured to receive the fourth signal from the allpass filter circuitry. The second combiner circuitry may be configured to generate an excursion limiting output signal.

Abstract: A sound volume controller includes a level adjustment section that adjusts a level of an input audio signal and outputs the adjusted input audio signal as an output audio signal, a level detection section that detects the level of the input audio signal, a volume value detection section that detects a volume value set by a user, a coefficient table storage section that stores a coefficient table which describes a relation between levels of the output audio signal and levels of the input audio signal, a coefficient table change section that changes the coefficient table stored in the coefficient table storage section in accordance with the volume value detected by the volume value detection section, and a setting section that sets an amount of level adjustment of the level adjustment section in accordance with the level of the input audio signal detected by the level detection section and the coefficient table.