Abstract: A noise attenuating system includes a core portion adapted to actively filter sound waves into various bands, passing only those bands corresponding to safe amplitude sounds to a wearer's ear canal. Unlike conventional active noise cancellation systems, active noise attenuation is accomplished without providing additional sound waves inverse to unsafe amplitude sound waves. Instead, unsafe amplitude sound waves are passively blocked, and only safe amplitude sound waves are passed through to the wearer's ear canal.

Abstract: A method and apparatus estimate additive noise in a noisy signal using incremental Bayes learning, where a time-varying noise prior distribution is assumed and hyperparameters (mean and variance) are updated recursively using an approximation for posterior computed at the preceding time step. The additive noise in time domain is represented in the log-spectrum or cepstrum domain before applying incremental Bayes learning. The results of both the mean and variance estimates for the noise for each of separate frames are used to perform speech feature enhancement in the same log-spectrum or cepstrum domain.

Abstract: A howling control apparatus and howling control method for controlling time up to the cancellation of howling suppression according to a howling occurrence situation, thereby eliminating the repetition of howling suppression and cancellation.

Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Abstract: In order to enhance the quality of a communication signal comprising speech signal components due to speech and noise signal components due to noise, a filter divides the communication signal into a plurality of frequency band signals representing the speech signal components and the noise signal components in a plurality of frequency bands. A calculator generates a plurality of weighting signals having weighting values corresponding to the frequency band signals. The weighting values represent at least approximations of the normalized powers of the noise signal components in the frequency band signals. The frequency band signals are altered in response to the weighting signals to generate weighted frequency band signals which are combined to generate a communication signal with enhanced quality.

Abstract: A multi-band spectral subtraction scheme is proposed, comprising a multi-band filter architecture, noise and signal power detection, and gain function for noise reduction. In one embodiment, the gain function for noise reduction consists of a gain scale function and a maximum attenuation function providing a predetermined amount of gain as a function of signal to noise ratio (“SNR”) and noise. In one embodiment, the gain scale function is a three-segment piecewise linear function, and the three piecewise linear sections of the gain scale function include a first section providing maximum expansion up to a first knee point for maximum noise reduction, a second section providing less expansion up to a second knee point for less noise reduction, and a third section providing minimum or no expansion for input signals with high SNR to minimize distortion.

Abstract: According to one embodiment of the present invention, a digital hearing device is disclosed. The digital hearing aid includes a microphone for receiving sound, which may include an analog signal. The analog signal is converted by a first converter into a digital signal. Filters are provided to divide the digital signal into multiple signal parts. A signal processor may be provided for each signal part, and performs signal processing on its respective signal part. An adder adds the output of the signal processors, which results in a processed digital signal. A second converter converts the processed digital signal back into an analog signal. A speaker then outputs the analog signal. According to another embodiment of the present invention, a method for enhancing sound is provided.

Abstract: The present invention relates to a hearing aid with an adaptive filter for suppression of acoustic feedback in the hearing aid. The hearing aid further comprises a controller that is adapted to compensate for acoustic feedback by determination of a first parameter of an acoustic feedback loop of the hearing aid and adjustment of a second parameter of the hearing aid in response to the first parameter whereby generation of undesired sounds is substantially avoided. Hereby a gain safety margin requirement is significantly reduced.

Abstract: The filtering coefficients of a frequency domain stereophonic echo canceller are adapted by a method which takes account of the cross-correlation between the input signals relating to the two channels. In particular, the adaptation process takes account of the coherence function, reducing the problems commonly encountered with stereophonic cancellation schemes when relatively correlated input signals occur.

Abstract: In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The AAC standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique.

Abstract: A method for utilizing validity constraints in a speech endpoint detector comprises a validity manager that may utilize a pulse width module to validate utterances that include a plurality of energy pulses during a certain time period. The validity manager also may utilize a minimum power module to ensure that speech energy below a pre-determined level is not classified as a valid utterance. In addition the validity manager may use a duration module to ensure that valid utterances fall within a specified duration. Finally, the validity manager may utilize a short-utterance minimum power module to specifically distinguish an utterance of short duration from background noise based on the energy level of the short utterance.

Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.

Abstract: A technique for realizing high-speed and high-precision acoustic reproduction using acoustic speakers. The audio signal processing system has a sub-band analysis filter bank that divides the entire frequency band of an input audio signal into multiple sub-bands. Filter coefficient calculation circuits identify equalizable sub-bands and compares equalizable sub-bands and corresponding sub-bands from output audio signals in order to calculate filter coefficients. A sub-band convolution filter bank and a filter circuit perform frequency convolution on calculated filter coefficients for equalizable sub-bands and process input audio signals on the basis of this convolution.

Abstract: An input signal is applied to a notch filter having a transfer function that is the inverse of the expected noise signal. The filtered signal is coupled to a first amplifier and the input signal is coupled to a second amplifier. The outputs of the amplifiers are summed. The gains of the amplifiers are oppositely adjusted in response to the magnitude of the input signal. At low amplitude, the filtered signal is amplified more than the unfiltered signal. At high amplitude, the unfiltered signal is amplified more than the filtered signal.

Abstract: A noise reduction apparatus (and methods). The apparatus has a housing and a processing device coupled to the housing. A sensor is coupled to the processing device and may be adapted to the housing. The sensor is adapted to determine a noise signal. A programmable memory is coupled to the processing device. The programmable memory device comprises 1 to N periodic frequency band limited noise wave shapes that are capable of reducing an intensity level of the noise signal. The apparatus also has an output device that is coupled to the processing device. The output device is adapted to output the periodic frequency band limited noise wave shape that is capable of reducing the intensity level of the noise signal.

Abstract: Methods and apparatus for blind channel estimation of a speech signal corrupted by a communication channel are provided. One method includes converting a noisy speech signal into either a cepstral representation or a log-spectral representation; estimating a correlation of the representation of the noisy speech signal; determining an average of the noisy speech signal; constructing and solving, subject to a minimization constraint, a system of linear equations utilizing a correlation structure of a clean speech training signal, the correlation of the representation of the noisy speech signal, and the average of the noisy speech signal; and selecting a sign of the solution of the system of linear equations to estimate an average clean speech signal in a processing window.

Abstract: One preferred embodiment of the present invention provides systems and methods for removing noise from an input analog signal in continues time. Briefly described in architecture, one embodiment of the system, among others, can be implemented as follows. An analog filtering system including VLSI circuitry separates an analog input signal into a plurality of sub-band signals. Then, an analog gain system including VLSI circuitry calculates a gain for each sub-band signal that suppresses the noise within the sub-band signal. In some preferred embodiments, VLSI circuitry includes floating gate technology. Methods and other systems are also provided.

Abstract: A sound field measuring apparatus has: an exponential pulse generator 11 which outputs a pulse signal to speakers 4a, 4b, . . . ; a microphone 6 which is disposed in an acoustic space 5 where the speakers 4a, 4b, . . . are disposed, and which detects a pulse signal output from each of the speakers 4a, 4b, . . . ; and a calculation section 15 which detects a time when the signal detected by the microphone 6 exceeds a predetermined threshold. The calculation section 15 calculates a time period from a time when the pulse signal is generated by the exponential pulse generator 11 to the time when the signal exceeds the predetermined threshold.

Abstract: A method for reducing interference in acoustic signals by using of an adaptive filter method involving spectral subtraction. The inventive method enables a significant reduction of interference in acoustic signals, especially voice signals, without causing any substantial falsification of said signals such as echo or musical tones, and significantly reduces computational requirements in comparison with other methods known per se that are similarly designed to improve signal quality.

Abstract: A sound producing system includes a loudspeaker to generate an audible signal. A transducer generates a pressure feedback signal from the audible signal. A controller produces a controlled signal for application to the loudspeaker. The controller derives the controlled signal from the pressure feedback signal and an audio input signal. The controlled signal improves the Sound Pressure Level (SPL) frequency response and reduces distortion within a selected bandwidth of the loudspeaker.

Abstract: An input signal is applied to a notch filter having a transfer function that is the inverse of the expected noise signal. The filtered signal is coupled to a first amplifier and the input signal is coupled to a second amplifier. The outputs of the amplifiers are summed. The gains of the amplifiers are oppositely adjusted in response to the magnitude of the input signal. At low amplitude, the filtered signal is amplified more than the unfiltered signal. At high amplitude, the unfiltered signal is amplified more than the filtered signal.

Abstract: A method and system for converting a sound signal containing a speech component and a noise component into recognizable language are disclosed, wherein the sound signal is transformed from a time domain into a frequency domain. Next the transformed signal is compared with a set of models of all possible sound signals to find a closest-matching known sound signal. A filter is then applied to the transformed signal. Here the filter corresponds to the model of the closest-matching known sound signal. Next a determination is made of an identity of the speech by searching a set of control data models to match a data model with the filtered transformed signal. Finally, a text stream representative of the determination is output, which enables a user not only to hear what may be a noisy voice message, but also to read the filtered message in some form, such as printed text or on a display screen.

Abstract: Method and apparatus for generating a reference signal includes parsing one or more input signals into a plurality of split signals in accordance with a predetermined criteria, generating split reference signals for each of at least two of the split signals, and combining a plurality of the split reference signals into a single reference signal. Multiple reference signals may be generated by mutiple pluralities of split reference signals. The predetermined criteria may include bandsplit filtering, and the output signals may be used for equalization, among other applications.

Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: A spectral subtraction is effected including: a first subtraction step in which overestimates of the spectral component of the noise are taken into account, to obtain spectral components of a first noise-suppressed signal; the computation of a masking curve by applying an auditory perception model on the basis of the spectral components of the first noise-suppressed signal; and a second subtraction step in which a respective quantity depending on parameters including a difference between the overestimate of the corresponding spectral component of the noise and the computed masking curve is subtracted from each spectral component of the speech signal in the frame. The result of the spectral subtraction is transformed into the time domain to construct a noise-suppressed speech signal.

Abstract: Method for eliminating spurious signal components (SS) from an input signal (ES), said method including the characterization, in a signal analysis phase (I), of the spurious signal components (SS) and of the information signal (NS) contained in the input signal (ES), and the determination or generation, in a signal processing phase (II), of the information signal (NS) or estimated information signal (NS′) on the basis of the characterization obtained in the signal analysis phase (I), said characterization of the signal components (SS, NS) being performed under utilization at least of auditory-based features (M1 to Mj).

Abstract: A system and method for removing noise from a signal containing speech (or a related, information carrying signal) and noise. A speech or voice activity detector (VAD) is provided for detecting whether speech signals are present in individual time frames of an input signal. The VAD comprises a speech detector that receives as input the input signal and examines the input signal in order to generate a plurality of statistics that represent characteristics indicative of the presence or absence of speech in a time frame of the input signal, and generates an output based on the plurality of statistics representing a likelihood of speech presence in a current time frame; and a state machine coupled to the speech detector and having a plurality of states. The state machine receives as input the output of the speech detector and transitions between the plurality of states based on a state at a previous time frame and the output of the speech detector for the current time frame.

Abstract: For tracking multiple, simultaneous voices, predicted tracking is used to follow individual voices through time, even when the voices are very similar in fundamental frequency. An acoustic waveform comprised of a group of voices is submitted to a frequency estimator, which may employ an average magnitude difference function (AMDF) calculation to determine the voice fundamental frequencies that are present for each voice. These frequency estimates are then used as input values to a recurrent neural network that tracks each of the frequencies by predicting the current fundamental frequency value for each voice present based on past fundamental frequency values in order to disambiguate any fundamental frequency trajectories that may be converging in frequency.

Abstract: The disclosed method uses the Wiener frequency filtering to suppress noise in noisy sound signals (u(t)). This method includes a preliminary step in which the sound signals (u(t)) to be noise-suppressed are digitized by sampling and subdivided into frames. The method then includes a first series of steps including the creation of a noise model on N frames, the estimating of the spectral density of the noise and of the energy of the noise model and the computing of a coefficient that reflects the statistical dispersion of the noise. It also includes a second series of steps including the computation of the spectral density of the signals to be noise-suppressed fore each frame. The coefficients of the Wiener filter are modified for each successively processed frame, by the parameters determined at the end of the two series of steps, so as to introduce an energy compensation and an adaptive overestimation of the noise.

Abstract: A noise or vibration control system reduces a sampling rate and reduces a control rate to improve computation efficiency. The present invention permits the use of a sample frequency (fs) that is less than twice the frequency of interest (fd). The sensed signals are filtered to extract a particular frequency range with a lower bound given by (2n−1)*fs/2 and an upper bound given by (2n+1)*fs/2, where n is an integer chosen so that the frequency of interest (fd) is within the extracted frequency range. The control commands are also calculated at a reduced rate, which is dependent upon the bandwidth of the tone, rather than the absolute frequency of the tone. Rather than updating the control signals directly on the sampled sensor data yk as it enters the computer, the control computations are done on the harmonic components ak and bk, or equivalently on the magnitude and phase.

Abstract: For the purpose of spatial reproduction of an audio signal, the latter must be projected onto the positions of the existing loudspeakers. It is desirable in this case not to have to be fixed on a specific loudspeaker configuration for transmitting the audio signal. However, a problem here is that a multiplicity of possible combinations exists. In the method according to the invention, the sound sources (3) are interpreted as acoustic objects for the purpose of projecting them onto an arbitrary loudspeaker configuration (2). Here, an acoustic object consists in that in addition to the audio signal a sound source is assigned an item of spatial information which specifies a virtual, spatial position of the sound source. In order to reproduce an acoustic object, the spatial information of the sound source and the actual position of a loudspeaker are used to calculate the virtual distance from the sound source via the loudspeaker to the hearer (1).

Abstract: A method and apparatus for enhancing a speech signal contaminated by additive noise through Kalman filtering. The speech is decomposed into subband speech signals by a multichannel analysis filter bank including bandpass filters and decimation filters. Each subband speech signal is converted into a sequence of voice frames. A plurality of low-order Kalman filters are respectively applied to filter each of the subband speech signals. The autoregression (AR) parameters which are required for each Kalman filter are estimated frame-by-frame by using a correlation subtraction method to estimate the autocorrelation function and solving the corresponding Yule-Walker equations for each of the subband speech signals, respectively. The filtered subband speech signals are then combined or synthesized by a multichannel synthesis filter bank including interpolation filters and bandpass filters, and the outputs of the multichannel synthesis filter bank are summed in an adder to produce the enhanced fullband speech signal.

Abstract: There is disclosed a method for processing a time-varying signal to produce a high-resolution spectrogram that represents power as a function of both frequency and time. Data blocks of a time series, which represents of a sampled signal, are subjected to processing which results in a sequence of frequency-dependent functions referred to as eigencoefficients. Each eigencoefficient represents signal information projected onto a local frequency domain using a respective one of K Slepian sequences or Slepian functions. The spectrogram is derived from time- and frequency-dependent expansions formed from the eigencoefficients.

Abstract: A digital audio coding apparatus includes a part which converts a frame of digital audio data into a frequency domain; a part which divides the digital audio data into a plurality of bands; a part which calculates an allowed distortion level by using an absolute hearing threshold for each divided band and assigns coding bits; a change part which changes the absolute hearing threshold adaptively on the basis of intensity distribution of the digital audio data in the frequency domain.

Abstract: Improved approaches are disclosed to filter and compress sound signals so as to achieve not only speech audibility and intelligibility at low levels but also preserves spectrum contrast at high levels. According to one aspect of the invention, gain amounts for different frequency bands are individually constrained based on signal levels for the frequency bands. Hence, the gain amounts for each of the frequency bands may or may not be constrained depending on the corresponding signal levels. As a result, the most critical information for speech intelligibility, speech clarity, and speech quality can be made available to hearing impaired people over wide range of signal level. The invention is particularly useful for hearing aids or other sound systems for the hearing impaired.

Abstract: With a view toward properly combining a fundamental echo with a harmonics echo according to the quality of a signal, the ratio between two frequency components of the fundamental echo is determined (704), and the ratio between two frequency components of the harmonics echo is determined (702). Further, a component ratio between a fundamental component and a harmonics component in an echo receive signal is adjusted based on these two ratios (706, 708).

Abstract: A noise suppress processing apparatus has a speech input section for detecting speech uttered by the speaker at different positions, an analyzer section for obtaining frequency components in units of channels by frequency-analyzing speech signals in units of speech detecting positions, a first beam former processor section for obtaining target speech components by suppressing noise in the speaker direction by filtering the frequency components in units of channels using filter coefficients, which are calculated to decrease the sensitivity levels in directions other than a desired direction, a second beam former processor section for obtaining noise components by suppressing the speech of the speaker by filtering the frequency components for the plural channels obtained by the analyzer section to set low sensitivity levels in directions other than a desired direction, an estimating section for estimating the noise direction from the filter coefficients of the first beam former processor section, and estimating

Abstract: A method and apparatus for processing a composite acoustic signal to reconstruct an acoustic signal that substantially matches a selected one of a plurality of sources. A plurality of microphones positioned at different spatial locations detect. variations in sound pressure level resulting from the activity of a plurality of acoustic sources at different locations. The outputs of the microphones are sampled and digitized, and the resulting digital waveform from each microphone is provided as an input to a corresponding filter bank. The outputs of the filter banks are input to a comparison unit. A comparison control unit generates “signature” information that characterizes each source with respect to the microphones. The comparison unit receives “signature” information of a selected source from the comparison control unit and provides an output to a synthesizer unit which produces a synthesized digital waveform for the selected source.

Abstract: A noise suppressor is provided which includes a signal to noise ratio (SNR) determiner, a channel gain determiner, a gain smoother and a multiplier. The SNR determiner determines the SNR per channel of the input signal. The channel gain determiner determines a channel gain &ggr;ch(i) per the ith channel. The gain smoother produces a smoothed gain {overscore (&ggr;ch+L (i,m))} per the ith channel and the multiplier multiplies each channel of the input signal by its associated smoothed gain {overscore (&ggr;ch+L (i,m))}.

Abstract: An audio signal processing device comprises signal supply means to supply over more than one input channel and per input channel over separate frequency subbands domain subchannels coded audio signals.

Abstract: An adaptive passive acoustic attenuation system implements control techniques to facilitate practical use of adaptive passive acoustic attenuation in industrial and commercial applications. The system includes multiple banks of multiple adjustable tuners that are used to passively attenuate an acoustic disturbance (e.g. a tone) propagating through an acoustic plant. All of the tuners in one of the banks are contemporaneously adjusted during an adaptation scan while the tuners in the other banks remain stationary. The process is then carried out for the other banks of adjustable tuners. The number of adjustable tuners per bank is chosen so that adaptation scans of the tuners in the respective bank create observable changes in acoustic levels to enable adaptation. Multiple sets of multiple banks of adjustable tuners can be provided to attenuate multiple disturbances propagating through the acoustic plant.

Abstract: A sound masking system for a multi-occupant work area includes a masking signal generator generating incoherent masking sound signals loudspeaker modules interconnected in a daisy-chain fashion, with each loudspeaker module receiving the masking sound signals on input connections and transmitting them to a successive loudspeaker module on output connections. The connections on which the masking sound signals appear in each loudspeaker are shifted by the inter-loudspeaker connections, such that successive loudspeakers automatically emit different masking sound signals for improved diffuseness in the overall masking sound in the work area. Each loudspeaker module has one jack having the input connections and another jack having the output connections, and each jack receives a detachable cable such as telephone cable to connect adjacent loudspeaker modules. The masking sound signals are shifted by a cross connection network between the two jacks in each loudspeaker module.

Abstract: A spectrum-based speech enhancement system estimates and tracks the noise spectrum of a mixed speech and noise signal. The system frames and windows a digitized signal and applies the frames to a fast Fourier transform processor to generate discrete Fourier transformed (DFT) signals representing the speech plus noise signal. The system calculates the power spectrum of each frame. The speech enhancement system employs a leaky integrator that is responsive to identified noise-only components of the signal. The leaky integrator has an adaptive time-constant which compensates for non-stationary environmental noise. In addition, the speech enhancement system identified noise-only intervals by using a technique that monitors the Teager energy of the signal. The transition between noise-only signals and speech plus noise signals is softened by being made non-binary. Once the noise spectrum has been estimated, it is used to generate gain factors that multiply the DFT signals to produce noise-reduced DFT signals.

Abstract: In correcting the sound field, the loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factors of the channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the delay times of the delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factor of the channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer, whereby the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: In correcting the sound field, the loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factors of the channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the delay times of the delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factor of the channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer, whereby the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: In correcting the sound field, loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11to ATFki for adjusting gains of the band-pass filters BPF11to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factors of channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The delay times of delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factor of a channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer. Therefore, the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: A device for analysis and filtration of sound which comprises at least one frequency-linear filter, at least one frequency-logarithmic filter and a weighting means for the combining and non-linear weighting of the output signals from the frequency-linear filter and the frequency-logarithmic filter. The sound is fed parallel as an input signal to the two sets of filters. In turn, the output signals from the filters are fed to the weighting means where they are combined and weighted non-linearly on the basis of magnitudes relevant to the sound. A decision with respect to the identity of the sound is made in a decision means. The invention also relates to a method for analyzing and filtering sound with the aid of the above-mentioned device.

Abstract: A personal sound masking system for use in an individual workspace provides an optimized acoustic background environment by delivering a sound masking signal that is specifically matched to the individual user's location and physical relationship to other nearby offices. The sound masking system employs multiple loudspeakers and multiple mutually incoherent channels in order to obtain a desired degree of diffuseness. A control module includes an erasable programmable read-only memory (EPROM) that stores data representing a number of samples of a masking signal segment, addressing logic that accesses the samples in the memory sequentially and repetitively to generate different series of data values each representing a different masking signal, digital to analog converters that convert the series of samples into analog masking signals, and power amplification circuitry that amplifies the analog masking signals to levels suitable for driving the loudspeakers.