Abstract: A signal processing device includes a bass signal extractor, a harmonic wave generator, a level detector, and an adjustment controller. The bass signal extractor first extracts a bass signal from an input audio signal. Natural-sounding bass enhancement is achieved as a result of the adjustment controller boosting the bass signal level until the level detector detects the bass signal level at a set level. For input bass signal levels higher than the set level, bass is enhanced virtually using a harmonic signal generated from the bass signal by the harmonic wave generator. As a result, the disadvantages of the boost method and the virtual signal enhancement method are mutually compensated for, and synergistic advantages for bass enhancement are obtained.

Abstract: A method of compensating for spatial audio frequency characteristics that varies in accordance with a mounting condition of a down firing speaker of an audio/video (AV) apparatus includes calculating a listening distance between the AV apparatus and a listener, calculating a distance between a speaker mounted on the AV apparatus and a neighboring reflective surface, setting a spatial frequency compensation filter value and a speaker frequency characteristic compensation filter value based on the calculated distances, and compensating for frequency characteristics of an audio signal by combining the spatial frequency compensation filter value and the speaker frequency characteristic compensation filter value.

Abstract: A device, system and method of playing back a digital audio stream wherein large amounts of pre-emphasis of the high frequencies is applied before the digital to analog conversion and before an interpolation or digital filter, followed by de-emphasis in the analog domain in order to yield better audio fidelity.

Abstract: The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Abstract: A processing apparatus which is suitable for signal communication with a system having at least one of an input module and an output module. The processing apparatus can be configured to receive input signals from the input module of the system. The processing apparatus includes a first channel processing portion to which the input signals are communicable. The first channel processing portion can be configurable to receive and process the input signals in a manner such that bass frequency audio signals are extracted from the input signals. The bass frequency audio signals can be further processed via at least one of linear dynamic range processing, manipulation of dynamic range via compression and manipulation of dynamic range via expansion.

Abstract: A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value ?, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on the response of an exponential transfer function.

Abstract: A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

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 signal processing apparatus includes a signal output unit for outputting a measurement signal, the measurement signal being produced by synthesizing a signal composed of a concatenation of 2d period signals with a sinusoidal signal, each period signal having a time-domain waveform period being 2n samples, the sinusoidal wave having a wave count within the concatenation period of 2d period signals being other than an integer multiple of 2d, and n and d being respectively natural numbers, and an analyzing unit for frequency analyzing a response signal obtained as a result of picking up the measurement signal output from the signal output unit.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: According to one embodiment, a sound corrector includes a signal outputter, a response signal, a frequency specifier, a coefficient specifier, a filter, and an outputter. The signal outputter outputs a measurement signal to measure acoustical properties of an object to be measured. The response signal receiver receives a response signal from the object in response to the measurement signal. The frequency specifier specifies a resonant frequency at a resonance peak from the response signal. The coefficient specifier specifies a correction coefficient of a correction filter for reducing the resonant frequency based on the specified resonant frequency. The filter performs filtering on a signal to be output to the object using the correction filter with the correction coefficient. The outputter outputs the signal having undergone the filtering to the object.

Abstract: Methods and systems for amplitude modulation in a parametric speaker system are provided that perform truncated double sideband (TDSB) frequency modulation of audio signal in which most of the processing is performed in the frequency domain, thus permitting use of fast processing techniques for amplitude modulation (AM) and filtering and reducing computation cost over time domain processing. A maximum envelope value of the time domain audio signal may be to the carrier signal in the frequency domain that avoids emitting the carrier signal when the input signal level is low or mute. The application of the envelope value may be smoothed to reduce discontinuity at input block boundaries.

Abstract: A system and method may be configured to reconstruct an audio signal from transformed audio information. The audio signal may be resynthesized based on individual harmonics and corresponding pitches determined from the transformed audio information. Noise may be subtracted from the transformed audio information by interpolating across peak points and across trough points of harmonic pitch paths through the transformed audio information, and subtracting values associated with the trough point interpolations from values associated with the peak point interpolations. Noise between harmonics of the sound may be suppressed in the transformed audio information by centering functions at individual harmonics in the transformed audio information, the functions serving to suppress noise between the harmonics.

Abstract: Method and arrangement in an audio handling entity, for damping of dominant frequencies in a time segment of an audio signal. A time segment of an audio signal is obtained, and an estimate of the spectral density or “spectrum” of the time segment is derived. An approximation of the estimate is derived by smoothing the estimate, and a frequency mask is derived by inverting the approximation. Frequencies comprised in the audio time segment are then damped based on the frequency mask. The method and arrangement involves no multi-band filtering or selection of attack and release times.

Abstract: Integrated processing of audio/video signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated system includes a main player that captures and processes signals digitally, a dummy display, and a dummy speaker. The dummy display may only have a display panel and a panel driver. The dummy speaker may only have a driving unit and no crossover logic. The main player may have a PC architecture and process all signals digitally for outputting signals tailored for the display device and the individual driving units of the dummy speaker. The integrated system may also provide dynamic signal adjustments based on the surrounding environment. The main player may include a storage device and can process media content stored therein to produce supplemental information to provide an optimal audiovisual experience. This supplemental information can be shared among users over a network connection.

Abstract: An audio processing device includes a) an input unit for converting a time domain input signal to a number NI of input frequency bands and b) an output unit for converting a number NO of output frequency bands to a time domain output signal. A signal processing unit processes the input signal in a number NP of processing channels, smaller than the number NI of input frequency bands. A frequency band allocation unit allocates input frequency bands to processing channels. A frequency band redistribution unit redistributes processing channels to output frequency bands, and a control unit dynamically controls the allocation of input frequency bands to processing channels and the redistribution of processing channels to output frequency bands.

Abstract: A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise.

Abstract: Bass frequencies of audio can be dynamically boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. Bass frequencies of audio can be boosted using a linear combination of an input audio signal and output of a high-pass filter. For example, bass frequencies of audio can be boosted by applying a high-pass filter to an input audio signal to produce an output of the high-pass filter, determining a current level, determining a target gain amount, dynamically adjusting the input audio signal and the output of the high-pass filter, and combining the gain-adjusted signals to produce an output signal. A dynamic bass boost filter can comprise a high-pass filter and a dynamic boost module.

Abstract: A system quantifies listener envelopment in a loudspeakers-room environment. The system includes a binaural detector that receives frequency modulated audible noise signals from multiple loudspeakers. The binaural detector generates detected signals that are analyzed to determine an objective listener envelopment. The envelopment is based on binaural activity of one or more sub-bands of the detected signal.

Abstract: Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S1) a set of features of the low band audio signal; mapping (S2) extracted features to at least one high band parameter with generalized additive modeling; frequency shifting (S3) a copy of the low band audio signal into the high band; controlling (S4) the envelope of the frequency shifted copy of the low band audio signal by said at least one high band parameter.

Abstract: A device may include a speaker, a vibration device and logic. The speaker may be configured to output audio signals. The logic may be configured to perform audio spectrum analysis associated with the audio signals, and synchronize output of the vibration device with the audio signals output by the speaker based on the audio spectrum analysis. The vibration device may also be configured to vibrate at a number of different frequencies based on the audio spectrum analysis.

Abstract: Disclosed are a method and apparatus for producing sounds in a virtual world, as well as a sound card. The method comprises the steps of determining a spring mass model of an object in the virtual world based on a 3D model of the object; analyzing force components produced on the 3D model by a collision in the virtual world; and generating sounds produced by the collision according to the spring mass model and the force components. By considering sound material of the object in the virtual world, the method, apparatus, and sound card may produce sounds in the virtual world more vividly and in real time.

Abstract: Methods and apparatus for tuning the audio path response of an audio device are described herein. The audio output of an audio device is captured and monitored over a predetermined frequency band. The frequency response of the captured audio output can be compared to one or more predetermined limits defining an acceptable frequency response. One or more dynamically configurable bands can be defined, and one or more parameters affecting the response within each configurable band can be adjusted. A simulated frequency response is produced and filter coefficients for a corresponding filter within the audio path are determined. The filter coefficients can be loaded to a digital filter within the audio path to modify the actual frequency response produced by the audio device.

Abstract: An audio processing apparatus is provided. A beamformer receives input signals and processes the input signals to generate a first processed signal. The input signals include at least one of a source signal and interference. A blocking matrix receives the input signals and operates to cancel the source signal from the input signals to generate a second processed signal. A first adaptive filter has adaptable first filter coefficients, generates a first filtered signal approximating the interference according to the first and second processed signals and continuously adapts the first filter coefficients according to the first filtered signal and the first processed signal. A second adaptive filter has adaptable second filter coefficients, generates a second filtered signal approximating the interference according to the first and second processed signals and selectively adapts the second filter coefficients according to the first filter coefficients and an output signal.

Abstract: A sound signal processing apparatus includes a frequency analysis unit which executes frequency analysis of an input sound signal; a low-frequency envelope calculating unit which calculates low-frequency envelope information as envelope information of a low-frequency band based on a result of the frequency analysis; a high-frequency envelope information estimating unit which applies learned data generated in advance based on a sound signal for learning and generates estimated high-frequency envelope information corresponding to an input signal from the low-frequency envelope information corresponding to the input sound signal; and a frequency synthesizing unit which synthesizes a high-frequency band signal corresponding to the estimated high-frequency envelope information generated by the high-frequency envelope information estimating unit with the input sound signal and generates an output sound signal in which a frequency band is expanded.

Abstract: The present invention relates to a method and corresponding system for predicting the perceived spatial quality of sound processing and reproducing equipment. According to the invention a device to be tested, a so-called device under test (DUT), is subjected to one or more test signals and the response of the device under test is provided to one or more means for deriving metrics, i.e. a higher-level representation of the raw data obtained from the device under test. The derived one or more metrics is/are provided to suitable predictor means that “translates” the objective measure provided by the one or more metrics to a predicted perceived spatial quality. To this end said predictor means is calibrated using listening tests carried out on real listeners. By means of the invention there is thus provided an “instrument” that can replace expensive and time consuming listening tests for instance during development of various audio processing or reproduction systems or methods.

Abstract: A signal processing system includes a level dependent bass management system. The level dependent bass management system utilizes audio input signal level information to apply at least one of multiple, available bass management solutions to generate one or more output signals from the audio input signal. In at least one embodiment, initially the level dependent bass management system boosts components of the audio input signal in the low frequency range by an amount sufficient to at least partially compensate for low frequency attenuation of the first speaker without exceeding one or more acceptable limitations of the signal processing system. If boosting alone cannot completely compensate for low frequency attenuation of the first speaker without exceeding one or more acceptable limitations of the signal processing system, the level dependent bass management system processes the audio input signal using an alternate low frequency management solution.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: An audio signal output apparatus includes a modulation signal generator for generating a first modulation signal by pulse-modulating an input audio signal of one channel using a first carrier signal or a first sampling clock, which has a first frequency; a vacuum tube filter unit comprising a vacuum tube and for generating a vacuum tube signal by allowing the first modulation signal to pass through the vacuum tube; a frequency modulation unit for generating a second modulation signal by pulse-modulating the vacuum tube signal; and a power switching amplifier for outputting an amplification signal corresponding to the second modulation signal.

Abstract: A sound element is generated by synthesizing, from a base sound, a frequency component of a sinusoidal wave one octave higher than the base sound. The base sound is a frequency component of one sinusoidal wave with an integer multiple of periods thereof matching a sample count represented by a power of 2. Sound elements having a frequency serving as a musical scale in a temperament are selected from the sound elements. The selected sound element is outputted in a predetermined pattern of time and a musical scale, so that a test tone is produced in a melody-like fashion.

Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: The present invention provides for methods and systems for digitally processing audio signals by adjusting the gain of a signal a first time, using a digital processing device located between a radio head unit and a speaker. The methods and systems may filter the adjusted signal with a first low shelf filter, compress the filtered signal with a first compressor, process the signal with a graphic equalizer, compress the processed signal with a second compressor and adjust the gain of the compressed signal a second time. The methods and systems may also filter the signal received from the first low shelf filter with a first high shelf filter prior to compressing the filtered signal with the first compressor. The signal may be filtered with a second low shelf filter prior to processing the signal with the graphic equalizer. Some embodiments filter the signal with a second high shelf filter after the signal is filtered with the second low shelf filter.

Abstract: Volume control circuits for use in electronic devices are provided. Some embodiments of the present invention provide volume control circuits for use in electronic devices. The volume control circuits include a gain circuit, a limiter circuit, a power measurement circuit and a gain adjustment circuit. The gain circuit is configured to receive an input audio signal and amplify the input audio signal based on a gain. The limiter circuit is configured to limit the amplified input audio signal based on a peak safety limit and output an output audio signal having a peak value based on the peak safety limit. The power measurement circuit is configured to measure a power level of the output audio signal. The power level has an associated power safety limit. The gain adjustment circuit is configured to adjust the gain of the gain circuit responsive to the measured power. Related methods and electronic devices are also provided herein.

Abstract: A method for transforming an audio signal and a system employing that method is disclosed. The method proceeds by extracting from the signal components that fall within a stop band, for example, using one or more low-pass filters. Then, a set of one or more harmonics of the extracted components is generated by down-sampling to create an intermediate signal, which is then repeated. Each harmonic is then weighted by controlling its gain by application to it of a respective weighting factor, the factors being determined by the psychoacoustic properties of a system that will reproduce the signal. Then, the weighted harmonics are added to the audio signal (which may have been subject to a delay to ensure synchronization) to create an output signal.

Abstract: A signal conditioner for subwoofers has a band pass filter which passes a lower frequency spectrum of an input signal. The band pass filter includes a high pass filter having a predetermined corner frequency, and a low pass filter having about the same corner frequency, and produces a filtered input to a peak voltage follower circuit. The peak voltage follower circuit traces and smoothes the upper boundary of the filtered input to produce a control voltage. An analog multiplier multiplies the input signal by the control voltage produced by the peak voltage follower to produce an enhanced signal where low frequency transients have been magnified.

Abstract: A system and method for minimizing the complex phase interaction between non-coincident subwoofer and satellite speakers for improved magnitude response control in a cross-over region. An all-pass filter is cascaded with bass-management filters in at least one filter channel, a[1d preferably all-pass filters are cascaded in each satellite speaker channel. Pole angles and magnitudes for the all-pass filters are recursively calculated to minimize phase incoherence. A step of selecting an optimal cross-over frequency may be performed in conjunction with the all-pass filtering, and is preferably used to select an optimal cross-over frequency prior to determining all-pass filter coefficients.

Abstract: Hough transform is performed on the point groups forming two dimensional data to generate a plurality of loci respectively corresponding to each of the point groups in a Hough voting space. When adding a voting value to a position in the Hough voting space through which the plurality of loci passes, addition is performed by varying the voting value based on a level difference between first and second signals respectively indicated by the two pieces of frequency decomposition information.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: The method comprises the steps of: filtering the audio signal by means of a lowpass filter (101) with a cutoff frequency substantially equal to said cutoff frequency (F0) of the sound playback device; determining a fundamental frequency for reconstituting from the lowpass filtered audio signal; and generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted. It also comprises the steps of: detecting a time envelope (env(t)) of the lowpass filtered audio signal; adapting the dynamic range of said time envelope (env(t)) as a function of the frequency band under consideration; and reinjecting said harmonic signal in phase into said audio signal by addition after multiplying said harmonic signal (Sharm) with the adapted time envelope (envadapt(t)).

Abstract: A sound and light control apparatus includes an audio signal input terminal for receiving an audio signal, a voltage dividing circuit electrically connected to the audio signal input terminal for dividing the voltage of the audio signal into two divided voltages, a control knob electrically connected to the voltage dividing circuit for adjusting the divided voltages, at least one audio signal output terminal electrically connected to the voltage dividing circuit and at least one loudspeaker for transmitting one divided voltage to the loudspeaker, a frequency-divider crossover circuit electrically connected to the voltage dividing circuit for filtering the other divided voltage according to a plurality of different audio frequencies to generate corresponding frequency division voltages, and transmitting each frequency division voltage to a corresponding frequency division voltage output terminal electrically connected to a light emitting diode and enabling the light emitting diode to produce light correspondin

Abstract: The perceived loudness of an audio signal is measured by modifying a spectral representation of an audio signal as a function of a reference spectral shape so that the spectral representation of the audio signal conforms more closely to the reference spectral shape, and determining the perceived loudness of the modified spectral representation of the audio signal.

Abstract: An audio signal processing apparatus includes: a division section that divides at least two or more channel audio signals into components in a plurality of frequency bands; a phase difference calculation section that calculates a phase difference between the two or more channel audio signals at each the frequency band; a level ratio calculation section that calculates a level ratio between the two or more channel audio signals at each the frequency band; a sound image localization estimation section that estimates, based on the level ratio or the phase difference, sound image localization at each the frequency band; and a control section that controls the estimated sound image localization at each the frequency band by adjusting the level ratio or the phase difference.

Abstract: A scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environment that has L sound sources located at prescribed positions in a prescribed room acoustics. The method provides a unified joint solution to the problems of equalizer design, crossover design, delay and level calibration, sum-response optimization and up-mixing. A multi-input multi-output audio precompensation controller is designed for an associated sound generating system including a limited number of loudspeaker inputs for emulating a number of virtual sound sources.

Abstract: A system and method for audio synthesizer utilizing frequency aperture cells (FAC) and frequency aperture arrays (FAA). In accordance with an embodiment, an audio processing system can be provided for the transformation of audio-band frequencies for musical and other purposes. In accordance with an embodiment, a single stream of mono, stereo, or multi-channel monophonic audio can be transformed into polyphonic music, based on a desired target musical note or set of multiple notes. At its core, the system utilizes an input waveform(s) (which can be either file-based or streamed) which is then fed into an array of filters, which are themselves optionally modulated, to generate a new synthesized audio output.

Abstract: A user selects a plurality of pieces of music data desired to be reproduced concurrently, at an input unit of an audio processing apparatus, from music data stored in a storage device. A reproducing apparatus reproduces selected music data respectively and generates a plurality of audio signals under the control of a control unit. An audio processing unit performs allocation of frequency band, extraction of a frequency component, time-division, periodic modulation, processing and allocation of a sound image, to respective audio signals under the control of the control unit. Then the audio processing unit attaches segregation information of audio signals and information on the degree of emphasis to respective audio signals. The down mixer mixes a plurality of audio signals and outputs as an audio signal having a predetermined number of channels, then an output unit outputs the signal as sounds.

Abstract: A method for increasing ring tone volume is provided. The method includes steps of: reading an audio file which is set as a current ring tone; determining whether the ring tone is a MP3 audio file or a musical instrument digital interface (MIDI) audio file; adjusting frequencies by using an equalizer technique to increase volume of the ring tone if the ring tone is the MP3 audio file; adjusting a volume level of the ring tone to be the highest volume level, and adjusting timbre of the ring tone to increase the ring tone volume by simulating a musical score of the ring tone by using different instruments if the ring tone is the MIDI audio file. A related system is also provided.

Abstract: An input audio signal having an input temporal envelope is converted into an output audio signal having an output temporal envelope. The input temporal envelope of the input audio signal is characterized. The input audio signal is processed to generate a processed audio signal, wherein the processing de-correlates the input audio signal. The processed audio signal is adjusted based on the characterized input temporal envelope to generate the output audio signal, wherein the output temporal envelope substantially matches the input temporal envelope.

Abstract: An apparatus and method for generating a binaural beat for brainwave induction is provided. Upon receipt of a stereo audio signal, a decorrelator extracts a common component existing in common in both channels of the stereo audio signal, and outputs the remaining stereo components except for the extracted common component. A frequency shifter generates a common component in which the binaural beat is included, by shifting a frequency of the extracted common component. First and second mixers mix the common component in which the binaural beat is included, with the remaining stereo components.

Abstract: A signal processing apparatus includes a plurality of equalizers configured to input an audio signal of a corresponding channel among audio signals of a plurality of channels and configured to perform at least gain adjustment on the basis of a set parameter, each of the equalizers being provided in such a manner as to correspond to an audio signal of one of the plurality of channels; a plurality of output sections configured to output each audio signal for each of the plurality of channels, the audio signal being processed by the equalizer; a measurement section configured to measure frequency-amplitude characteristics of the audio signal output from the output section; and a computation section configured to perform a computation process for correcting frequency-amplitude characteristics of an audio signal of each channel on the basis of the measurement result by the measurement section.