Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: Bass frequencies of audio can be boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, bass frequencies of audio can be boosted using an integer bass boost filter by receiving user-settable parameters, such as “c” and “s” coefficients, and implementing the integer bass boost filter using a coupled form structure implementation and the user-settable parameters. Bass frequencies of audio can also be boosted using an integer bass boost filter that is configured to use any of plural coupled form structure implementations. Bass frequencies of audio can be also be boosted using a linear combination of an input audio signal and output of a high-pass filter.

Abstract: For hearing apparatus and especially for hearing devices a tone balance unit is to be able to be used even under difficult general conditions. Therefore a hearing apparatus with a signal processing device, through which signals are able to be processed in a number of frequency channels, and a control device for controlling the level of the individual frequency channels are provided. The setting device features an unsymmetrical tone balance unit related to a predetermined frequency with which at least one lower and at least one upper channel of the number of frequency channels related to a predetermined frequency is able to be simultaneously unsymmetrically adjusted relative to the predetermined frequency. Thus for example a tone balance unit can also be used with a low feedback limit in the upper frequency range.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.

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

Abstract: In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events and/or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Abstract: There is provided an automatic gain control circuit for volume control receiving at least one audio signal input via either a wired connection or a wireless connection. The circuit includes an output signal regulator receiving a first portion of the at least one audio signal input, with the regulator being able to vary an amplitude of the first portion of the at least one audio signal input when a control signal from a circuit controller is received. The circuit controller may be powered by a DC generator; with the DC generator receiving a second portion of the at least one audio signal input. A signal from the output signal regulator may be passed through an acoustic generator. It is preferable that the first portion of the at least one audio signal input is substantially more than the second portion. It is also preferable that the output signal regulator varies the amplitude of the at least one audio signal input by reducing the amplitude. A corresponding method of volume control is also disclosed.

Abstract: A device and method for improving the performance of an electro-acoustic transducer. An acoustic test signal is generated through the electro-acoustic transducer. The acoustic test signal is measured at multiple points on an ambient surface around the electro-acoustic transducer to create measured acoustic data. Based on the measured acoustic data, an acoustic power frequency response of the electro-acoustic transducer is calculated. A correction impulse response for the electro-acoustic transducer is determined based on the acoustic power frequency response. A correction filter applies the correction impulse response to a sound signal input to generate a sound signal output for playback through the electro-acoustic transducer.

Abstract: An audio processing apparatus is disclosed which includes: a first signal generation portion configured to generate an audio signal of which the frequency is varied over time; an operation portion; a storage portion configured to store characteristic information in accordance with the frequency and an amplitude of the audio signal in effect when the operation portion is operated; a reproduction portion configured to reproduce audio data; and a correction portion configured to correct a reproduced signal from the reproduction portion based on the characteristic information stored in the storage portion.

Abstract: A speaker device comprises a first speaker (SP1) for reproducing an audio signal and a second speaker (SP2) adapted for reproducing an audio signal and spaced from the first speaker horizontally by a predetermined distance. At least one of the first and second speakers phase has varying means (APF1, APF2, . . . ) for varying the phase of the audio signal by a predetermined quantity of phase according to (i) the frequency of the audio signal and (ii) the predetermined distance.

Abstract: A control method to automatically adjust the volume of a sound transmitter based on the measurement and the computation of the acoustic statistics of the room where the sound is emitted. The system comprises at least one sensor, a calculator determining statistics of the signal collected by the at least one sensor and a controller using these statistics provided by the calculator to adjust the volume of the transmitter in the room.

Abstract: A handheld wireless communications device has an adjustable volume setting. The communications device also has a downlink audio processor containing a digital audio filter. Storage in the device contains a first set of filter coefficients that configure the filter to enable the device to pass a wireless communications device certification process while sacrificing sound quality, at a low volume setting. The storage also contains a second set of filter coefficients that configure the filter to enhance sound quality while sacrificing passing the wireless communications device certification process, at a high volume setting. Other embodiments are also described and claimed.

Abstract: An apparatus and method of automatically compensating audio volume in response to a channel change. The method includes generating a gain value to adjust a level of an input audio signal by comparing an average level of the input audio signal with a predetermined input/output level curve, adjusting an audio volume of the input audio signal by applying the generated gain value to the level of the input audio signal, and adjusting a dynamic range of the audio signal having the adjusted audio volume.

Abstract: Loudness-based compensation systems and techniques are described which provide audio compensation in noisy environments. Loudness approximations are determined for an audio block both by itself and in the presence of background noise. These approximations utilize compression of audio intensity within frequency bands in order to better reflect listeners' hearing perception. From these loudness approximations, a gain is determined for the audio block and then applied in such a manner that the effect is not jarring to listeners.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: According to one embodiment, during mixing of an N-channel input signal to generate an M-channel output signal, in at least one frequency sub-band, magnitude equalization is applied to the mixed channel signals such that an amplitude sum magnitude for the N input channels (e.g., the magnitude of a sum of estimated amplitudes of the N input channels) is approximately equal to an amplitude sum magnitude for the M output channels (e.g., the magnitude of a sum of estimated amplitudes of the M output channels). In one implementation, magnitude equalization is applied to one or more sub-bands (e.g., those below 1 kHz), and power equalization is applied to one or more other sub-bands (e.g., those above 1 kHz) to reduce coloration effects in the output signal.

Abstract: An audio signal processing apparatus includes a harmonic overtone adder and an equalizer. The harmonic overtone adder includes a high-pass filter for extracting from an audio signal higher than a first predetermined frequency, a filter for extracting a frequency component lower than half a second predetermined frequency, an harmonic overtone generator for generating a frequency-doubled harmonic overtone component from an output from the filter, and a first combining unit for combining the frequency component output from the high-pass filter and the harmonic overtone component output from the harmonic overtone generator.

Abstract: An apparatus is provided for controlling a graphic equalizer which is implemented by digital filters, each being assignable to a frequency band for regulating a level of the frequency band. In the apparatus, a set of control devices are provided in correspondence to respective frequency bands, each control device being operable to specify the level of the corresponding frequency band to either of a reference level or other level than the reference level. An assignment section operates when a number of the digital filters is less than a total number of the frequency bands, for assigning the digital filters to the frequency bands which are specified with the levels other than the reference level. A control section controls remaining digital filters which are not assigned to any of the frequency bands in a through state.

Abstract: A system and method for enhancement and management of streaming audio is disclosed. In one embodiment, the system provides a client-side decoder that is compatible with numerous audio formats, so that a user can enjoy relatively high-quality audio from various sources, even from sources that do not provide multi-channel or high-quality audio data. The system and method also include a management system for managing and controlling the use of licensed signal processing software to further enhance an audio stream. In one embodiment, the management system is used to manage a signal processing module that provides psychoacoustic audio processing to create a wider soundstage, an acoustic correction process to increase the perceived height and clarity of the audio image, and bass enhancement processing to create the perception of low bass from the small speakers or headphones typically used with multi-media systems and portable audio players.

Abstract: A request is received to play an audio file. A determination is made regarding whether volume normalization parameters associated with the audio file are stored in a media library. If the volume normalization parameters associated with the audio file are stored in the media library, the volume normalization parameters are retrieved from the media library. If the volume normalization parameters associated with the audio file are not stored in the media library, retrieving the volume normalization parameters from the audio file. The volume normalization parameters are applied while playing the audio file. The volume normalization process can be applied across multiple audio files during playback.

Abstract: In a multichannel amplifier according to the present invention, in a case where a headphone is not connected to a headphone output terminal, a switch IC and first to third switches are on/off controlled by a control unit so that an audio signal path for outputting a front channel signal FL/FR to a front speaker and extending from a first input terminal 1 to a front speaker output terminal 6 via a buffer circuit 10a, a switch IC 9, a first volume control unit 3a, a buffer circuit 10b, and a first amplifying unit 4a is not electrically connected to another audio signal path.

Abstract: A method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

Abstract: LPF and HPF extract bass and treble ranges, respectively, from an input sound signal, and bass and treble boost circuits perform dynamic range expansion/contraction on the extracted bass- and treble-range sound signals in accordance with input levels of the sound signals. The input sound signal and the sound signals output from the boost circuits are added together. There may also be provided coefficient calculation sections for calculating filter coefficients on the basis of the levels of the sound signals extracted by the LPF and HPF. In this case, the bass and treble boost sections perform, in accordance with the filter coefficients calculated by the corresponding coefficient calculation sections, filter processes for increasing/decreasing the levels of the bass and treble ranges, respectively.

Abstract: A volume control apparatus comprises signal processing section for performing digital signal processing and a scaling process on an audio signal outputted from a sound source; level adjusting section for amplifying or attenuating a level of the audio signal outputted from the signal processing section, by an amount of adjustment determined based on an instructed amount of change, and outputting the amplified or attenuated audio signal to an outside; storage section for storing a level value of the audio signal outputted to the outside and the amount of adjustment for the audio signal adjusted by the level adjusting section; detecting section for detecting, when a volume controller is controlled by a user, time required for an amount of operation to reach a predetermined value; first setting section for setting an amount of operational change based on the level value stored in the storage section and the detected time; second setting section for setting an allowable amount of change by which the level adjustin

Abstract: A system and method for graphic equalization of audio signals is disclosed. Traditional graphic equalizers provide control over the gains in each of a set of frequency bands. However, the actual band gains vary from the desired gains due to crosstalk between bands. Prior art methods for addressing this difficulty include applying a correction filter to the equalizer, and adjusting the shape of the individual band filters, both of which increase the computational cost. In an embodiment of the present invention, the input gains are processed to produce a set of adjusted gains which take into account the crosstalk, and result in an equalization interpolating the input band gains.

Abstract: Systems, devices, and methods are provided to inhibit apparent amplitude modulation in non-linear processing that causes distortion in a processed signal. One aspect of the invention includes a hearing aid. The hearing aid includes a microphone to receive an input signal, a speaker to reproduce the input signal, and a processor. The processor processes the input signal using a gain. The processor includes an inhibitor, which inhibits distortions, and an adjuster, which adjusts the gain. The inhibitor acts to smooth an envelope of the input signal to inhibit undesired modulation. The adjuster adjusts the gain if the envelope is either above or below a threshold. The hearing aid further includes a compression recapture system to supply the compressed portion of the input signal to more closely reproduce the actual input signal.

Abstract: A display device of the present invention is mounted on an automotive vehicle. An obstacle located in front of the vehicle is detected by a camera, and its features including speed, distance, size, etc. are analyzed by electronic circuits. An image showing the features of the front object is outputted from a liquid crystal panel. The outputted image is reflected on the windshield, displaying a virtual image to a driver. The virtual image is displayed in a form of a spot encircling the front obstacle seen through the windshield. The ways of displaying the encircling spot are variously changed according to a degree of collision danger that is determined according to a period in which the vehicle reaches the obstacle. For example, if the degree of collision danger is high, the spot is shown in red-orange color and/or with a high brightness. Thus, the driver easily recognizes the front obstacle in a high degree of collision danger.

Abstract: A circuit configured to adjust the level of an audio signal includes filters each of which is configured to receive an audio signal, and to pass a band set for the filter. Variable gain amplifiers are severally provided to the respective filters, and each variable gain amplifier amplifies the output signal of the corresponding filter. An adder sums the input audio signal and the output signals of the variable gain amplifiers. A control unit controls the level of the output signal of each of the variable gain amplifiers, and controls the gain of each variable gain amplifier according to the level thus monitored.

Abstract: There is provided an apparatus for enabling karaoke. The apparatus includes a casing for the apparatus; a microphone array incorporated within the casing, the microphone array being concealed from a user by the casing; a controller coupled to the microphone array to at least process audio signals from the user input at the microphone array; a data storage device coupled to the controller, the data storage device being for storing pre-recorded songs used for karaoke, and for storing songs as sung by the user; and an image capturing device coupled to the controller, the image capturing device being for capturing images of the user while the user is singing, with the captured images of the user being stored on the data storage device.

Abstract: A method and apparatus for processing an audio signal in multiple audio frequency bands while minimizing undesirable changes in tonal qualities of the audio signal by determining an initial gain adjustment factor for each audio frequency band resulting from the application of an audio processing technique. A final gain adjustment factor for each band is selected from a corresponding set of weighted or unweighted initial gain adjustment factors. The set of initial gain adjustment factors from which the final gain adjustment factor for a specified audio frequency band is obtained is derived from other audio frequency bands that have the frequency of the specified band as a harmonic frequency. Changes in audio signal level within one audio frequency band thereby affect the signal level of harmonic frequencies to decrease relative changes in volume between a fundamental frequency and its harmonics.

Abstract: An audio output apparatus has a measuring circuit which measures the levels of at least two sound signals, a sound level adjusting module (a sound level adjusting circuit and a gain control circuit) which adjusts a sound level so as to equal the levels of the sound signals based on the levels measured at the measuring circuit, and an array speaker unit (a delay circuit, a multiplier, an adder, an amplifier and a speaker unit) which emits sounds in accordance with the sound signals outputted from the sound level adjusting module in different directivities respectively.

Abstract: An audio compensation unit includes a detection subunit and a compensation subunit. The detection subunit is for detecting an audio signal to obtain an original frequency. The detection subunit defines a predetermined low frequency band and a predetermined high frequency band. The detection subunit is also for generating a first control signal when the original frequency is within the low frequency band, and a second control signal when the original frequency is within the high frequency band. The compensation subunit includes an electronic switch, a first compensation circuit, and a second compensation circuit. The electronic switch enables the first compensation circuit according to the first control signal, for compensating the audio signal. The electronic switch is further enables the second compensation circuit according to the second control signal, for compensating the audio signal.

Abstract: To provide a speaker array apparatus and a method for setting audio beams in a speaker array apparatus, in which the degree of freedom in the place where the speaker array apparatus is installed is high, and a user can set audio beams easily. A speaker array apparatus 1 sweeps a range of from 0 degree to 180 degrees in front of a speaker array 10 with audio beams based on an audio signal limited to a band where the angles of the audio beams can be adjusted. The speaker array apparatus 1 collects direct sounds or reflected sounds of the audio beams through a nondirectional microphone 2. The speaker array apparatus 1 analyzes the collected audio data, detects peaks not lower than a threshold value, and checks symmetry among the peaks. When there is a symmetry, the angles where the peaks were detected are set as angles with which audio beams of respective channels of a surround-sound should be output.

Abstract: A receiving apparatus comprising: a frequency conversion unit configured to convert a received radio frequency signal to an intermediate frequency signal; an automatic gain control unit configured to control an amplitude level of at least either one of the radio frequency signal and the intermediate frequency signal according to a gain control signal; a demodulation unit configured to demodulate an audio signal from the intermediate frequency signal; a correction level output unit configured to output a correction level signal corresponding to a difference between a demodulation unit input level and a predetermined reference level, the demodulation unit input level being an amplitude level of the intermediate frequency signal inputted to the demodulation unit; and an addition unit configured to add the gain control signal and the correction level signal, to be outputted as a signal strength signal indicating received signal strength of the radio frequency signal.

Abstract: The invention discloses a method and a circuit for preventing clipping of an audio signal. The method comprises determining the highest amplitude frequency components of the audio signal in the frequency domain, and then reducing the amplitudes of these frequency components until a level of clipping of the audio signal falls below a predetermined level.

Abstract: A sound reproducing system for use with multiple rooms includes a main control unit, and at least one secondary control unit. Thus, the respective automatic level control circuit maintains the voice signal output at a constant level, the respective power amplification circuit of the main control unit converts the voice signal into a larger voice signal, the volume control circuit of the respective secondary control unit converts the larger voice signal and the noise produced during transmission into a smaller voice signal, and the power amplification circuit of the respective secondary control unit converts the smaller voice signal into a larger voice signal output, thereby producing a voice signal output with a higher quality.

Abstract: A variable gain amplifier amplifies an input audio signal. A first control unit monitors the level of the input audio signal, and sets the gain of the variable gain amplifier to a predetermined fixed value when this level is lower than a predetermined minimum threshold. A second control unit monitors the level of an output signal VO of the variable gain amplifier. When this level exceeds a predetermined maximum threshold, the second control unit lowers the gain of the variable gain amplifier. When this level becomes smaller than this threshold, it raises the gain. When the level of the input audio signal is smaller than the minimum threshold, the first control unit disables the gain control operation of the variable gain amplifier controlled by the second control unit. A sub-amplifier amplifies the input audio signal. An output mixing unit mixes signals output from the variable gain amplifier and the sub-amplifier.

Abstract: A tone emphasizing device includes a tone emphasizing unit that includes a level detecting unit that detects a fundamental tone level of a tone of a given pitch included in an original tone reproduced by a data player; an extracting unit that extracts frequency components of a fundamental tone and overtones from the original tone; a frequency correcting unit that eliminates, from the extracted frequency components, frequency components equal to or lower than the fundamental tone and frequency components equal to or higher than an overtone included in the tone of the given pitch; and a level adjusting unit that adjusts, according to results obtained by the level detecting unit, a level of the corrected tone. The tone emphasizing device further includes an amplifying unit that amplifies to a desired level, the tone having an adjusted level; and an adding unit that adds to the original tone, the amplified tone.

Abstract: In an apparatus for improving voice clarity by controlling a gain of a voice based on a sound pressure level of the voice and a sound pressure level of noise, it is determined whether a sound pressure level of a gain-controlled voice exceeds a maximum allowable level. If the sound pressure level exceeding the maximum allowable level is caused by audio sound, a sound pressure level of the audio sound is reduced. The sound pressure level of the audio sound is minimally reduced so that the sound pressure level of the voice becomes equal to or lower than the maximum allowable level.

Abstract: A system and method for dynamic bass frequency control, the system comprising: a bandwidth filter coupled to a plurality of switchable electronic components for filtering a frequency range of an input signal; an amplifier coupled to the bandwidth filter and the plurality of switchable electronic components, the amplifier being configured for adjusting amplitude of the filtered frequency range of the input signal based on a configuration of the plurality of switchable electronic components; and a signal level detector coupled to the switchable electronic components for determining signal strength of the input signal, the signal level detector being configured to switch the plurality of switchable electronic components between more than one configuration based on the signal strength of the input signal, each configuration determining the frequency range of the input signal to be filtered by the bandwidth filter and the amplitude adjustment to be made to the filtered frequency range of the input signal by the am

Abstract: A control signal is generated by an operating means for being operated by a driver. In response to this control signal, an input signal to rear-seat headphones is temporarily attenuated or is replaced by another given signal or a transmission signal from the driver.

Abstract: An audio signal (A) is enhanced by dividing the signal into time segments of a selected frequency range and scaling the audio signal in each time segment. The time segments (S) are defined by zero crossings (Z) of the audio signal, thus avoiding the introduction of any undesired harmonics. The scaling may involve linear or non-linear scaling factors. When the selected frequency range comprises bass frequencies, a very effective and distortion-free bass enhancement is obtained.

Abstract: According to one embodiment, a video/audio output method according to one embodiment includes detecting a playback volume value set in an external speaker controller for controlling an external speaker, calculating a decibel value corresponding to the detected playback volume value, generating an audio signal corresponding to the calculated decibel value, and outputting the generated audio signal to the external speaker controller, and, at the same time, outputting a video based on a video signal corresponding to the audio signal.

Abstract: Disclosed are a circuit device which prevents an over-output of an audio signal in a certain band, a processing method of an audio signal, an image processing apparatus including the same and a control method thereof The image processing apparatus including: an image processor which processes an image signal; and an audio processor which filters an input audio signal to be in a first frequency band and limits a level of the audio signal in the first frequency band to a first output threshold value or less if the level of the audio signal in the first frequency band exceeds a predetermined first threshold value.

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 level adjustment method is applicable to an audio processing apparatus having a plurality of amplifiers corresponding to three or more of input channels of an audio signal for amplifying signal levels of the respective input channels. The level adjustment method is carried out by a group arrangement process of arranging the plurality of the input channels into one or more group, and a group control process of controlling each group such as to decrease an amplification rate of all the amplifiers corresponding to the input channels belonging to the same group as a maximal one of the signal levels of the input channels belonging to the same group increases. Typically, the group arrangement process is applied to an audio signal of a surround system having at least six input channels including a left channel, a right channel, a left surround channel, a right surround channel, a center channel and an LFE channel.

Abstract: A method for audio distortion processing is provided, whereby narrowband and wideband gains are incrementally reduced and recovered for controlling audio distortion of an audio reproduction system. Reduction limits are determined for both the narrowband and wideband gains, as a function of operator controlled inputs. If clipping is detected, the narrowband gain is initially reduced until either the narrowband gain reduction limit has been achieved or the clipping desists. The wideband gain is subsequently reduced if the reduction limit of the narrowband gain has been achieved and the clipping persists. The wideband gain is reduced until either the wideband gain reduction limit has been achieved or the clipping desists. After the clipping desists, a gain recovery process ensues, whereby the wideband gain is initially recovered until at its original level and the narrowband gain is subsequently recovered until reaching its original level.

Abstract: In a device equipped with an audio output, a range of the audio output level in regular use is set. When an attempt is made to change the setting to a volume level exceeding the range of the audio output level having been set, a request is made for an acknowledgement that the operation is appropriate, and only if a reply is received within a given time, is the volume setting changed. Such a method prevents the audio output from a television set or audio equipment to be increased to a volume level higher than necessary due to an operating error in a volume setting operation.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: In a sound signal processing device, the level of an output signal is controlled as a function of the level of the corresponding input signal by supplying the output signal, whose level is converted by a level converter and to which a specific frequency characteristic is added, to an automatic level control section. The amplification factor of the signal is automatically controlled so as to prevent the output signal level from exceeding an allowable range. The sound signal processing device includes a control section that variably changes the gain values of two level converters and the threshold value, which is a parameter to be used by a gain computer in the automatic level control section, as a function of the peak gain that is a defined value of an equalizer as defined by the user by way of an operation section.