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258 results about "Psychoacoustics" patented technology
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Psychoacoustics is the scientific study of sound perception and audiology—how humans perceive various sounds. More specifically, it is the branch of science studying the psychological and physiological responses associated with sound (including noise, speech and music). It can be further categorized as a branch of psychophysics. Psychoacoustics received its name from a field within psychology—i.e., recognition science—which deals with all kinds of human perceptions. It is an interdisciplinary field of many areas, including psychology, acoustics, electronic engineering, physics, biology, physiology, and computer science.
Multichannel audio system having audio channel compensation
ActiveUS9100766B2Uniform levelTwo-channel systemsLoudspeaker spatial/constructional arrangementsTarget ResponseVocal tract
Owner:HARMAN INT IND INC
Apparatus and method for coding a time-discrete audio signal and apparatus and method for decoding coded audio data
A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded / decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder / decoder structure results.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
ActiveUS8090120B2Reduce the differenceReduce impactGain controlSpeech analysisFrequency spectrumEqualization
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.
Owner:DOLBY LAB LICENSING CORP
Scalable compressed audio bit stream and codec using a hierarchical filterbank and multichannel joint coding
ActiveUS20070063877A1Facilitate efficient scalingSuitable for useSpeech analysisCode conversionComputer hardwareData rate
A method for compressing audio input signals to form a master bit stream that can be scaled to form a scaled bit stream having an arbitrarily prescribed data rate. A hierarchical filterbank decomposes the input signal into a multi-resolution time / frequency representation from which the encoder can efficiently extract both tonal and residual components. The components are ranked and then quantized with reference to the same masking function or different psychoacoustic criteria. The selected tonal components are suitably encoded using differential coding extended to multichannel audio. The time-sample and scale factor components that make up the residual components are encoded using joint channel coding (JCC) extended to multichannel audio. A decoder uses an inverse hierarchical filterbank to reconstruct the audio signals from the tonal and residual components in the scaled bit stream.
Owner:DTS
Adaptive filter for speech enhancement in a noisy environment
InactiveUS7117145B1Improve clarityImprove easeSpeech analysisDigital computer detailsEnvironmental noiseAdaptive filter
A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes a microphone for receiving the spoken voice and the ambient noise and for converting the spoken voice and the ambient noise into an audio signal, the audio signal having a first component corresponding to the spoken voice and a second component corresponding to the ambient noise, a speech enhancement filter for removing the second component from the audio signal to provide a filtered audio signal, the speech enhancement filter removing the second component by processing the audio signal by a method taking into account elements of psycho-acoustics of a human ear, and a loudspeaker for outputting a clarified voice in response to the filtered audio signal.
Owner:LEAR CORP
Audio coding device with fast algorithm for determining quantization step sizes based on psycho-acoustic model
ActiveUS20060074693A1Reduce the amount of calculationSpeech analysisFast algorithmHuman auditory system
An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors. A coder encodes at least one of the quantized values, common scalefactor, and individual scalefactors.
Owner:FUJITSU LTD
Method for executing automatic evaluation of transmission quality of audio signals using source/received-signal spectral covariance
InactiveUS6651041B1Increase valueHigh signal energy of signalSpeech analysisWireless communicationObjective assessmentCovariance
A source signal (e.g. a speech sample) is processed or transmitted by a speech coder 1 and converted into a reception signal (coded speech signal). The source and reception signals are separately subjected to preprocessing 2 and psychoacoustic modelling 3. This is followed by a distance calculation 4, which assesses the similarity of the signals. Lastly, an MOS calculation is carried out in order to obtain a result comparable with human evaluation. According to the invention, in order to assess the transmission quality a spectral similarity value is determined which is based on calculation of the covariance of the spectra of the source signal and reception signal and division of the covariance by the standard deviations of the two said spectra.The method makes it possible to obtain an objective assessment (speech quality prediction) while taking the human auditory process into account.
Owner:ASCOM
Sound intelligibility enhancement using a psychoacoustic model and an oversampled filterbank
InactiveUS7050966B2Improve signal qualityImproving signal signal intelligibilityEar treatmentNoise generationSignal of interestEngineering
A system and method of improving signal intelligibility over an interference signal is provided. The system includes a psychoacoustic professor having a psychoacoustic model wherein the level of a signal-of-interest is improved so as to be audible above noise and so as not to exceed a predetermined maximum output level. The system can be combined with active noise cancellation.
Owner:HIMPP
Amplitude-scaling resilient audio watermarking method and apparatus based on quantization
InactiveUS20050043830A1User identity/authority verificationSpeech analysisIntermediate frequencyAmplitude scaling
Disclosed is an amplitude-scaling resilient audio watermarking apparatus and method. An encoding apparatus includes: a polyphase filterbank for dividing an audio signal into a plurality of subbands; a psychoacoustic module for applying a psychoacoustic model to the audio signal to provide a signal-to-mask ratio; a watermark encoder for evaluating an encoding parameter from the subbands according to the signal-to-mask ratio and embedding the encoding parameter and a watermark into subbands corresponding to the middle frequency; and a synthesis filterbank for synthesizing the subband signals to output a watermarked audio signal. A decoding apparatus includes: a polyphase filterbank for dividing a received audio signal into the predetermined number of subbands; an EM estimator for estimating an scale factor from an encoding parameter contained in the audio signal and a watermarked subband according to an EM algorithm, and generating the quantizer step size Δd of a decoder according to the amplitude-scaling; a watermark decoder for extracting a watermark from a subband corresponding to the middle frequency considering the quantizer step size; and an integrated determiner for integrating outputs of the watermark decoder to determine a watermark.
Owner:ELECTRONICS & TELECOMM RESINST
Method for objectively and quantifiably evaluating noise fret degree in vehicle based on auditory model
InactiveCN101672690AImprove performanceOvercoming complexityVibration measurement in fluidPsychotechnic devicesHuman bodyProcess mechanism
The invention relates to a method for objectively and quantifiably evaluating noise fret degree in a vehicle based on an auditory model, in particular to an evaluation method of psychoacoustics and vehicle sound quality. The method comprises the following steps: designing a dummy head model; imitating a processing mechanism of the middle ear and the inner ear of a human body to the sound by usingan auditory peripheral calculation model; collecting vehicle noise signals under the working conditions of uniform, accelerated and idle speed; pre-treating the noise sample and processing the loudness in specification; obtaining a psychoacoustics parameter by using a sound quality calculation model; obtaining subjective evaluation result test data by using a grouped and paired comparison method;calculating and painting a correlated scatter plot chart between each parameter and the ranking value of the subjective evaluation result; and analyzing and calculating to obtain the objectively quantifiable model of the subjective fret degree under each working condition. The invention can realize the psychoacoustic parameter calculation of objectively evaluating the sound quality with differentvehicle types, gears and speeds, wherein the calculated value has good pertinence and consistency with the evaluated result of the subjective evaluation method. The invention has stable evaluated result and high reliability, and can improve the sound quality and competitiveness of vehicles combined with the design of new CAE cars.
Owner:JILIN UNIV
Feedforward prediction of scalefactors based on allowable distortion for noise shaping in psychoacoustic-based compression
A method of encoding a digital signal, particularly an audio signal, which predicts favorable scalefactors for different frequency subbands of the signal. Distortion thresholds which are associated with each of the frequency subbands of the signal are used, along with transform coefficients, to calculate total scaling values, one for each of the frequency subbands, such that the product of a transform coefficient for a given subband with its respective total scaling value is less than a corresponding one of the distortion thresholds. In an audio encoding application, the distortion thresholds are based on psychoacoustic masking. The invention may use a novel approximation for calculating the total scaling values, which obtains a first term based on a corresponding distortion threshold, and obtains a second term based on a sum of the transform coefficients. Both of these terms may be obtained using lookup tables. The total scaling values can be normalized to yield scalefactors by identifying one of the total scaling values as a minimum nonzero value, and using that minimum nonzero value to carry out normalization. Encoding of the signal further includes the steps of setting a global gain factor to this minimum nonzero value, and quantizing the transform coefficients using the global gain factor and the scalefactors.
Owner:CIRRUS LOGIC INC
Time domain dynamic characteristic analysis-based intra-vehicle sound quality prediction method
InactiveCN105021275AFeature reservedReduce boundary effectsSubsonic/sonic/ultrasonic wave measurementTime domainAnalysis study
The invention belongs to the technical field of sound quality prediction in vehicle vibration and noise analysis and control and relates to a time domain dynamic characteristic analysis-based intra-vehicle sound quality prediction method. With the method adopted, the problem of incapability of current intra-vehicle sound quality analysis researches to describe the time domain dynamic characteristics of intra-vehicle sound environment in vehicle regular driving can be solved. The method includes the following steps that: 1, a sound acquisition device is utilized to acquire an intra-vehicle sound signal sample; 2, framing and windowing processing is performed on the intra-vehicle sound signal sample; 3, a subjective evaluation test is performed on the intra-vehicle sound signal sample; 4, sound quality objective psychological acoustic parameters are calculated; 5, the time domain dynamic characteristic indexes of the objective psychological acoustic parameters are calculated, wherein the indexes include the variances of the objective psychological acoustic parameters and the ranges of the objective psychological acoustic parameters; 6, an intra-vehicle sound quality objective comprehensive evaluation model is established; 7, the output value of the sound quality objective comprehensive evaluation model is compared with a subjective evaluation value; and 8, a test sample is utilized to verify the model.
Owner:JILIN UNIV
Method and System for Ultrasonic Signaling, Ranging and Location Tracking
ActiveUS20140192622A1Minimize audible artifactUser privacy is enhancedBeacon systems using ultrasonic/sonic/infrasonic wavesDirection finders using ultrasonic/sonic/infrasonic wavesMultilaterationFrequency modulation
An indoor ultrasonic location tracking system that can utilize standard audio speakers to provide indoor ranging information to modern mobile devices like smartphones and tablets. The method uses a communication scheme based on linearly increasing frequency modulated chirps in the audio bandwidth just above the human hearing frequency range where mobile devices are still sensitive. The method uses gradual frequency and amplitude changes that minimize human perceivable (psychoacoustic) artifacts derived from the non-ideal impulse response of audio speakers. Chirps also benefit from Pulse Compression, which improves ranging resolution and resilience to both Doppler shifts and multi-path propagation that plague indoor environments. The method supports the decoding of multiple unique identifier packets simultaneously. A Time-Difference-of-Arrival pseudo-ranging technique allows for localization without explicit synchronization with the broadcasting infrastructure. An alternate received signal strength indicator based localization technique allows less accurate localization at the benefit of sparser transmission infrastructure.
Owner:CARNEGIE MELLON UNIV
Front surround system and method of reproducing sound using psychoacoustic models
InactiveUS20070286427A1Enhanced Stereo EffectImprove system performanceMicrophonesLoudspeakersIntermediate frequencyLoudspeaker
A front surround reproduction system improving the stereo effect of mid and low frequency signals by using a psychoacoustic model, and a method thereof. An audio reproducing system to reproduce multi-channel audio signals by using a plurality of speakers includes a split unit to copy the input multi-channel signals and to split the signals into two groups of multi-channel signals, a virtual sound processing unit to generate a virtual sound signal based on a head related transfer function (HRTF) from the one group of the multi-channel signals split in the split unit, a beam forming processing unit to generate a sound beam signal by adjusting the delays and levels of the multi-channel signals belonging to the other group split in the split unit, and a crossover network unit to adjust the characteristics of the virtual sound signal and the sound beam signal generated in the virtual sound processing unit and the beam forming processing unit, respectively, and to provide the virtual sound signal and the sound beam signal to mid and low frequency speaker arrays and high frequency speaker arrays, respectively.
Owner:SAMSUNG ELECTRONICS CO LTD
Method of encoding an audio signal using a quality value for bit allocation
A method for encoding an audio signal, including providing a masking function, representative of psychoacoustic masking; setting a quality value for data of the encoded signal, adjusting the masking function dependent upon the quality value; and allocating bits for quantization of the encoded signal based on the incremental masking function.
Owner:STMICROELECTRONICS ASIA PACIFIC PTE
Apparatus and method for coding a time-discrete audio signal to obtain coded audio data and for decoding coded audio data
A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded / decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder / decoder structure results.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Adaptive environmental noise compensation for audio playback
InactiveUS20110251704A1Prevent maskingSignal processing for reducing noiseSpeech analysisEnvironmental noiseNoise level
The present invention counterbalances background noise by applying dynamic equalization. A psychoacoustic model representing the perception of masking effects of background noise relative to a desired foreground soundtrack is used to accurately counterbalance background noise. A microphone samples what the listener is hearing and separates the desired soundtrack from the interfering noise. The signal and noise components are analyzed from a psychoacoustic perspective and the soundtrack is equalized such that the frequencies that were originally masked are unmasked. Subsequently, the listener may hear the soundtrack over the noise. Using this process the EQ can continuously adapt to the background noise level without any interaction from the listener and only when required. When the background noise subsides, the EQ adapts back to its original level and the user does not experience unnecessarily high loudness levels.
Owner:DTS
Methods and apparatus for maximizing speech intelligibility in quiet or noisy backgrounds
ActiveUS20050114127A1Improve speech clarityMaximize intelligibility metricSpeech recognitionPublic address systemHearing test
Methods and apparatus for maximizing speech intelligibility use psycho-acoustic variables of a model of speech perception to control the determination of optimal frequency-band specific gain adjustments. Speech signals (or other audio input) whose intelligibility is to be improved are characterized by parameters which are applied to the model. These include measurements or estimates of speech intensity level, average noise spectrum of the incoming audio signal, and / or the current frequency-gain characteristic of the hearing compensation device. Characterizations of listeners based on hearing test results, for example, may also be applied to the model. Frequency-band specific gain adjustments generated by use of the model can be used for hearing aids, assistive listening devices, telephones, cellular telephones, or other speech delivery systems, personal music delivery systems, public-address systems, sound systems, speech generating systems, or other devices or mediums which project, transfer or assist in the detection or recognition of speech.
Owner:ARTICULATION
Spatially constant surround sound system
ActiveUS20120243713A1Desirable balanceLoudspeaker spatial/constructional arrangementsStereophonic systemsVirtual userPsychoacoustics
An audio processing system may modify an input surround sound signal to generate a spatially equilibrated output surround sound signal that is perceived by a user as spatially constant for different sound pressures of the surround sound signal. The audio processing system may determine based on a psychoacoustic model of human hearing, a loudness and a localisation for a combined sound signal. The loudness and the localisation may be determined by the system for a virtual user located between the front and the rear loudspeakers that has a predetermined head position in which one ear of the virtual user is directed towards one of front or rear loudspeakers and the other ear of the virtual user being directed towards the other of the front or rear loudspeakers. The audio processing system may adapt the front and / or rear audio signal channels based on the determined loudness and localisation.
Owner:HARMAN BECKER AUTOMOTIVE SYST
Speech Enhancement Employing a Perceptual Model
ActiveUS20100076769A1Reduce gainAdditional componentSpeech recognitionSpeech synthesisMusical toneAcoustic model
Speech enhancement based on a psycho-acoustic model is disclosed that is capable of preserving the fidelity of speech while sufficiently suppressing noise including the processing artifact known as “musical noise”.
Owner:DOLBY LAB LICENSING CORP
Scalable compressed audio bit stream and codec using a hierarchical filterbank and multichannel joint coding
ActiveUS7548853B2Effectively scaledSuitable for useSpeech analysisCode conversionComputer hardwareDifferential coding
A method for compressing audio input signals to form a master bit stream that can be scaled to form a scaled bit stream having an arbitrarily prescribed data rate. A hierarchical filterbank decomposes the input signal into a multi-resolution time / frequency representation from which the encoder can efficiently extract both tonal and residual components. The components are ranked and then quantized with reference to the same masking function or different psychoacoustic criteria. The selected tonal components are suitably encoded using differential coding extended to multichannel audio. The time-sample and scale factor components that make up the residual components are encoded using joint channel coding (JCC) extended to multichannel audio. A decoder uses an inverse hierarchical filterbank to reconstruct the audio signals from the tonal and residual components in the scaled bit stream.
Owner:DTS
System for providing a reduction of audiable noise perception for a human user
InactiveUS20090074199A1Reduce or even eliminate the human perception of an auditory noiseReduce perceptionEar treatmentNoise generationNoise controlPattern perception
A system is disclosed where the well-known psycho-acoustic masking effect, i.e. frequency and / or temporal masking is applied to reduce the human perception of a noise signal. An input signal, such as music or another entertainment signal, is adjusted based on the intensity of the auditory noise by applying existing knowledge about the properties of the human auditory perception and is provided to the human user as a masking sound signal, so that the masking sound elevates the human auditory perception threshold for at least some of the noise signal, whereby the user's perception of that part of the noise signal is reduced or eliminated. The masking sound may be combined with active noise control where a sound in anti-phase with the noise signal is provided for further reduction of the human perception of the noise signal.
Owner:MAYSOUND
Audio coding device with fast algorithm for determining quantization step sizes based on psycho-acoustic model
An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors. A coder encodes at least one of the quantized values, common scalefactor, and individual scalefactors.
Owner:FUJITSU LTD
Cellular automata music generator
The Cellular Automata Music Generator (CAMG) is a software platform, application and engine that generates and modifies musical compositions in real-time using a system of networked modules that utilize the 1-dimensional cellular automata (CA) mathematical model. The music compositions are non-looping (i.e. the same few bars of music do not repeat over and over), can be any length, can have low memory and processing requirements compared to looping WAV or MP3 files, and follow psychoacoustic principles. The music generated is ever evolving and can be globally altered based on one-parameter change. The music engine can be utilized in computer games or to generate unique ringtones or any musical composition for cell phones or other mobile devices based on ten-digit inputs, such as telephone numbers. The ringtones or musical pieces can be generated automatically based on the telephone number of the in-coming telephone call, or manually by the user, who inputs a selected numerical sequence. It can also be used as an audio caller identification system.
Owner:CONDITION30
Energy dependent quantization for efficient coding of spatial audio parameters
ActiveUS20070016416A1Quantized more efficientlyEfficiently quantizedSpeech analysisCode conversionEnergy measurePsychoacoustics
Parameters being a measure for a characteristic of a channel or of a pair of channels, wherein the parameter is a measure for a characteristic of the channel or of the pair of channels with respect to another channel of a multi-channel signal can be quantized more efficiently using a quantization rule that is generated based on a relation of an energy measure of the channel or the pair of channels and an energy measure of the multi-channel signal. With generation of the quantization rule taking into account a psycho acoustic approach, the size of an encoded representation of the multi-channel signal can be decreased by coarser quantization without significantly disturbing the perceptual quality of the multi-channel signal when reconstructed from the encoded representation.
Owner:DOLBY INT AB +2
Audio signal encoding method, program of audio signal encoding method, recording medium having program of audio signal encoding method recorded thereon, and audio signal encoding device
InactiveUS20070244699A1Reduce loadReduce processing loadSpeech analysisCode conversionSignal onPsychoacoustics
Disclosed herein is an audio signal encoding method for dividing an audio signal into a plurality of subband signals, allocating bits to the subband signals on a basis of psychoacoustic analysis, and encoding the audio signal, in which the audio signal encoding method may include an evaluation criterion calculating step, a sorting step and a repetitive process step.
Owner:SONY CORP
Systems and methods for high resolution signal analysis and chaotic data compression
Systems and methods for processing, compressing, and distributing data, such as an audio file, are provided. Single- and multi-channel data streams are transformed into a single signal in a Unified Domain. A high resolution frequency analysis based on phase evolution provides accurate frequency estimates and distinguishes between oscillatory and noise-like signal components. The unified signal components are then prioritized using a Psychoacoustic Model. The prioritized components can be arranged in layers based on the component priorities and compressed (e.g., with a chaotic compression scheme). The least psychoacoustically important layers can be removed to lower the transmission bitrate. Digital rights management tools based, for example, on a unique device identification can be used for secure distribution.
Owner:GROOVE MOBILE
System and method for enhanced streaming audio
InactiveUS7277767B2Increase valueEnhancement of the audio streamMultiple digital computer combinationsTwo-channel systemsHeadphonesLoudspeaker
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.
Owner:DTS
Cellular Automata Music Generator
The Cellular Automata Music Generator (CAMG) is a software platform, application and engine that generates and modifies musical compositions in real-time using a system of networked modules that utilize the 1-dimensional cellular automata (CA) mathematical model. The music compositions are non-looping (i.e. the same few bars of music do not repeat over and over), can be any length, can have low memory and processing requirements compared to looping WAV or MP3 files, and follow psychoacoustic principles. The music generated is ever evolving and can be globally altered based on one-parameter change. The music engine can be utilized in computer games or to generate unique ringtones or any musical composition for cell phones or other mobile devices based on ten-digit inputs, such as telephone numbers. The ringtones or musical pieces can be generated automatically based on the telephone number of the in-coming telephone call, or manually by the user, who inputs a selected numerical sequence. It can also be used as an audio caller identification system.
Owner:CONDITION30
Automatic loudspeaker equalization method
ActiveCN103634726AImprove performanceImprove sound reproduction abilityStereophonic systemsTransducer circuitsHarmonicEqualization
The invention provides an automatic loudspeaker equalization system, particularly relates to an automatic loudspeaker equalization method, and aims to improve sound replaying performance of a loudspeaker system on full bands. The method includes measuring pulse responses at one or more position points in a room by a microphone to acquire a frequency response of each position point and low-frequency lower limit of the loudspeaker system, and acquiring an equilibrium filter by the adaptive optimization algorithm to compensate the loudspeaker system. For low-frequency signals with frequency lower than the lower-limit frequency of the loudspeaker system, ultraharmonics components of fundamental-frequency signals are generated according to the psychoacoustics based fundamental-frequency missing principle and are superposed with delayed original sound signals after being subjected to gain control, and accordingly, the sound replaying performance of the loudspeaker system on full bands is improved.
Owner:SUZHOU SONAVOX ELECTRONICS
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